From deveshbissa at rediffmail.com Mon Oct 1 04:15:13 2007 From: deveshbissa at rediffmail.com (devesh bissa) Date: 1 Oct 2007 08:15:13 -0000 Subject: [SIPForum-discussion] border controller for sip Message-ID: <20071001081513.32247.qmail@f5mail-237-201.rediffmail.com> Hi, I want to use border controller(openSBC) with IMS network for sip.Please help (how to configure and use it) if anyone did it previously.Thank youDevesh -------------- next part -------------- An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071001/2ed6efb3/attachment.html From kit_del_rosario at yahoo.com Mon Oct 1 04:54:52 2007 From: kit_del_rosario at yahoo.com (Francisco del rosario) Date: Mon, 1 Oct 2007 01:54:52 -0700 (PDT) Subject: [SIPForum-discussion] ASTERISK PBX INTEGRATION WITH A SOFTSWITCH Message-ID: <923711.18892.qm@web52011.mail.re2.yahoo.com> Hi, Need your input to get the right script for Asterisk PBX. The issue is for outgoing calls when connected to a mobile device. The call is connected but there is no speech path i.e. no voice received from both ends. Any advice to provide the right script file . ? Thanks... ____________________________________________________________________________________ Moody friends. Drama queens. Your life? Nope! - their life, your story. Play Sims Stories at Yahoo! Games. http://sims.yahoo.com/ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071001/854ec241/attachment.html From j.scholz at teles.de Mon Oct 1 12:58:09 2007 From: j.scholz at teles.de (Joerg Scholz) Date: Mon, 1 Oct 2007 18:58:09 +0200 Subject: [SIPForum-discussion] specific case with a=inactive Message-ID: Hello everybody, I have a quick question related to the SDP signaling behavior in a specific hold case. My scenario is the following: Incoming Re-invite of an established voice call with the following content of SDP: c=IN IP4 0.0.0.0 ... a=inactive In the 200ok I confirm the attribute line: a=inactive and the call is hold - everything ok. Later on comes another Re-invite without SDP for the same call. Now my question, if I answer in the 200ok for that Re-invite with SDP do I have to repeat also the: a=inactive attribute or do I send now my SDP again with a=sendrecv Thanks and Best regards Joerg -------------- next part -------------- An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071001/ddadd9c4/attachment.html From vivian_cyn at hotmail.com Mon Oct 1 13:03:01 2007 From: vivian_cyn at hotmail.com (ChenVivian) Date: Mon, 1 Oct 2007 17:03:01 +0000 Subject: [SIPForum-discussion] 3pcc implementation with sip provider Message-ID: Hi, My question might be very simple since I am new to SIP. The problem is that I have implemented a third part call controller following flow IV in rfc3725. I tested it using P2P module, it works fine. But when I applied it to sip server, sip.voipstunt.com, which kept sending me the 4XX message (bad request or unsupported media type) on response to my INVITE(no SDP). I don't know what could cause the failure. I was wondering if it is because the sip server doesn't support 3pcc messages. should both sip server and gateway support 3pcc in this situation? Thanks very much. Best Regards,

Yuening Chen

Department of Computer Science,

Uppsala University,

Sweden.

vivian_cyn at hotmail.com

_________________________________________________________________ Windows Live Spaces ???????????????? http://miaomiaogarden2007.spaces.live.com/ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071001/372ddd8c/attachment.html From rjsparks at nostrum.com Mon Oct 1 16:26:45 2007 From: rjsparks at nostrum.com (Robert Sparks) Date: Mon, 1 Oct 2007 15:26:45 -0500 Subject: [SIPForum-discussion] SIPit 21 registration closes Oct 29 Message-ID: <1EC7DF16-E27B-4E6F-AEB5-DEA4C284E7C4@nostrum.com> SIPit 21 will take place November 5 through 9, 2007 in Beijing, China. Registration will close October 29 (four weeks from now). If you haven't already registered, please reserve your seat now before the event fills up. This SIPit will be hosted by the BII Group and the Beijing University of Posts and Telecommunications. The registration fee is $575 US Dollars per participant. See http://www.sipit.net for more information and to register. See you in Beijing! RjS From robert.traussnig at kapsch.net Tue Oct 2 01:20:34 2007 From: robert.traussnig at kapsch.net (Traussnig Robert) Date: Tue, 2 Oct 2007 07:20:34 +0200 Subject: [SIPForum-discussion] specific case with a=inactive In-Reply-To: Message-ID: <23D52CF27A66BF4E9352313D60E46F3E0113EA3D@EXCLUSTER.kcc.local> Hi! I'm not quite sure if this is correct but because of the fact that you didn't get a SDP in the Re-Invite you have to give an offer in your 200ok. So you have to decide if you send a=inactive again or a=sendrecv. Maybe someone have a better answer. Best Regards, Robert -----Ursprungliche Nachricht----- Von: discussion-bounces at sipforum.org [mailto:discussion-bounces at sipforum.org]Im Auftrag von Joerg Scholz Gesendet: Montag, 01. Oktober 2007 18:58 An: discussion at sipforum.org Betreff: [SIPForum-discussion] specific case with a=inactive Hello everybody, I have a quick question related to the SDP signaling behavior in a specific hold case. My scenario is the following: Incoming Re-invite of an established voice call with the following content of SDP: c=IN IP4 0.0.0.0 ... a=inactive In the 200ok I confirm the attribute line: a=inactive and the call is hold - everything ok. Later on comes another Re-invite without SDP for the same call. Now my question, if I answer in the 200ok for that Re-invite with SDP do I have to repeat also the: a=inactive attribute or do I send now my SDP again with a=sendrecv Thanks and Best regards Joerg -------------- next part -------------- An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071002/8d631d7b/attachment.html From kunusan at yahoo.com Tue Oct 2 04:40:57 2007 From: kunusan at yahoo.com (badal naik) Date: Tue, 2 Oct 2007 01:40:57 -0700 (PDT) Subject: [SIPForum-discussion] specific case with a=inactive In-Reply-To: <23D52CF27A66BF4E9352313D60E46F3E0113EA3D@EXCLUSTER.kcc.local> Message-ID: <288012.81621.qm@web56704.mail.re3.yahoo.com> Well, I think in your scenario U have to add the SDP parameter of the previous success(2**) reponse. Re-Invite should work as follows in ur case: When a UAC sends a re-invite with no session description, in which case the first reliable non-failure response to the re-invite will contain the offer. That is my understanding.. Thanks Badal Naik --- Traussnig Robert wrote: > Hi! > > I'm not quite sure if this is correct but because of > the fact that you > didn't get a SDP in the Re-Invite you have to give > an offer in your > 200ok. So you have to decide if you send a=inactive > again or a=sendrecv. > Maybe someone have a better answer. > > Best Regards, > Robert > > > -----Ursprungliche Nachricht----- > Von: discussion-bounces at sipforum.org > [mailto:discussion-bounces at sipforum.org]Im Auftrag > von Joerg Scholz > Gesendet: Montag, 01. Oktober 2007 18:58 > An: discussion at sipforum.org > Betreff: [SIPForum-discussion] specific case with > a=inactive > > > > Hello everybody, > > I have a quick question related to the SDP signaling > behavior in a > specific hold case. My scenario is the following: > > > > Incoming Re-invite of an established voice call with > the following > content of SDP: > > c=IN IP4 0.0.0.0 > > ... > > a=inactive > > > > In the 200ok I confirm the attribute line: > > a=inactive > > > > and the call is hold - everything ok. > > Later on comes another Re-invite without SDP for the > same call. Now my > question, if I answer in the 200ok for that > Re-invite with SDP do I have > to repeat also the: > > a=inactive > > attribute or do I send now my SDP again with > > a=sendrecv > > > > Thanks and Best regards > > Joerg > > > > > _______________________________________________ > This is the SIP Forum discussion mailing list > TO UNSUBSCRIBE, or edit your delivery options, > please visit > http://sipforum.org/mailman/listinfo/discussion > Post to the list at discussion at sipforum.org > ____________________________________________________________________________________ Yahoo! oneSearch: Finally, mobile search that gives answers, not web links. http://mobile.yahoo.com/mobileweb/onesearch?refer=1ONXIC From j.scholz at teles.de Tue Oct 2 05:05:08 2007 From: j.scholz at teles.de (Joerg Scholz) Date: Tue, 2 Oct 2007 11:05:08 +0200 Subject: [SIPForum-discussion] specific case with a=inactive Message-ID: Thanks Badel and Robert for your responds. That's also what I expected to do - but it seems to be a problem for that class 5 SIP server solution which I try to work with. It uses the Re-invite without SDP to retrieve the hold call. Thereby the retrieve is initiated with a user agent web application (not the phone). If I send still: A=inactive In the SDP of the 200ok response of my device; it is used for the second leg of the call as SDP content and the call will be reestablished without voice. So I need exactly to know what the right behavior is in that case. It is quite simple to change the behavior but I'm pretty sure that other cases will have problem then. Thanks again and best regards Joerg -----Original Message----- From: badal naik [mailto:kunusan at yahoo.com] Sent: Tuesday, October 02, 2007 10:41 AM To: Traussnig Robert; Joerg Scholz; discussion at sipforum.org Subject: Re: [SIPForum-discussion] specific case with a=inactive Well, I think in your scenario U have to add the SDP parameter of the previous success(2**) reponse. Re-Invite should work as follows in ur case: When a UAC sends a re-invite with no session description, in which case the first reliable non-failure response to the re-invite will contain the offer. That is my understanding.. Thanks Badal Naik --- Traussnig Robert wrote: > Hi! > > I'm not quite sure if this is correct but because of > the fact that you > didn't get a SDP in the Re-Invite you have to give > an offer in your > 200ok. So you have to decide if you send a=inactive > again or a=sendrecv. > Maybe someone have a better answer. > > Best Regards, > Robert > > > -----Ursprungliche Nachricht----- > Von: discussion-bounces at sipforum.org > [mailto:discussion-bounces at sipforum.org]Im Auftrag > von Joerg Scholz > Gesendet: Montag, 01. Oktober 2007 18:58 > An: discussion at sipforum.org > Betreff: [SIPForum-discussion] specific case with > a=inactive > > > > Hello everybody, > > I have a quick question related to the SDP signaling > behavior in a > specific hold case. My scenario is the following: > > > > Incoming Re-invite of an established voice call with > the following > content of SDP: > > c=IN IP4 0.0.0.0 > > ... > > a=inactive > > > > In the 200ok I confirm the attribute line: > > a=inactive > > > > and the call is hold - everything ok. > > Later on comes another Re-invite without SDP for the > same call. Now my > question, if I answer in the 200ok for that > Re-invite with SDP do I have > to repeat also the: > > a=inactive > > attribute or do I send now my SDP again with > > a=sendrecv > > > > Thanks and Best regards > > Joerg > > > > > _______________________________________________ > This is the SIP Forum discussion mailing list > TO UNSUBSCRIBE, or edit your delivery options, > please visit > http://sipforum.org/mailman/listinfo/discussion > Post to the list at discussion at sipforum.org > ____________________________________________________________________________ ________ Yahoo! oneSearch: Finally, mobile search that gives answers, not web links. http://mobile.yahoo.com/mobileweb/onesearch?refer=1ONXIC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071002/d9a1f360/attachment-0001.html From kunusan at yahoo.com Tue Oct 2 07:10:25 2007 From: kunusan at yahoo.com (badal naik) Date: Tue, 2 Oct 2007 04:10:25 -0700 (PDT) Subject: [SIPForum-discussion] specific case with a=inactive In-Reply-To: Message-ID: <858677.33596.qm@web56704.mail.re3.yahoo.com> Hello Joerg, I am not sure whether this will be helpful to you.But take it as a suggestion. As i Know call hold is implemented in two ways. 1. Send a re-Invite with "0.0.0.0" as the IP address in the sdp data.(RFC 2543) 2.Send a re-Invite with the parameter a=sendonly set in the sdp data.(RFC 3264 ) You have tried the first one.Can u check the authenticity of the second one. In the first case the recepient cant send RTCP to you.I think most advanced server uses RFC 3264 for call Hold.That may be the case that ur server expects this RFC and you are following the obsolete one. Try with second one.I think it may work.. But please let me know if things work out. Thanks Badal Naik --- Joerg Scholz wrote: > Thanks Badel and Robert > for your responds. That's also what I expected to do > - but it seems to be a > problem for that class 5 SIP server solution which I > try to work with. > It uses the Re-invite without SDP to retrieve the > hold call. Thereby the > retrieve is initiated with a user agent web > application (not the phone). If > I send still: > A=inactive > In the SDP of the 200ok response of my device; it is > used for the second leg > of the call as SDP content and the call will be > reestablished without voice. > > So I need exactly to know what the right behavior is > in that case. > > It is quite simple to change the behavior but I'm > pretty sure that other > cases will have problem then. > Thanks again and best regards > Joerg > > > -----Original Message----- > From: badal naik [mailto:kunusan at yahoo.com] > Sent: Tuesday, October 02, 2007 10:41 AM > To: Traussnig Robert; Joerg Scholz; > discussion at sipforum.org > Subject: Re: [SIPForum-discussion] specific case > with a=inactive > > Well, > I think in your scenario U have to add the SDP > parameter of the previous success(2**) reponse. > Re-Invite should work as follows in ur case: > When a UAC sends a re-invite with no session > description, in which case the first reliable > non-failure response to the re-invite will contain > the > offer. > > That is my understanding.. > > Thanks > Badal Naik > --- Traussnig Robert > wrote: > > > Hi! > > > > I'm not quite sure if this is correct but because > of > > the fact that you > > didn't get a SDP in the Re-Invite you have to give > > an offer in your > > 200ok. So you have to decide if you send > a=inactive > > again or a=sendrecv. > > Maybe someone have a better answer. > > > > Best Regards, > > Robert > > > > > > -----Ursprungliche Nachricht----- > > Von: discussion-bounces at sipforum.org > > [mailto:discussion-bounces at sipforum.org]Im Auftrag > > von Joerg Scholz > > Gesendet: Montag, 01. Oktober 2007 18:58 > > An: discussion at sipforum.org > > Betreff: [SIPForum-discussion] specific case with > > a=inactive > > > > > > > > Hello everybody, > > > > I have a quick question related to the SDP > signaling > > behavior in a > > specific hold case. My scenario is the following: > > > > > > > > Incoming Re-invite of an established voice call > with > > the following > > content of SDP: > > > > c=IN IP4 0.0.0.0 > > > > ... > > > > a=inactive > > > > > > > > In the 200ok I confirm the attribute line: > > > > a=inactive > > > > > > > > and the call is hold - everything ok. > > > > Later on comes another Re-invite without SDP for > the > > same call. Now my > > question, if I answer in the 200ok for that > > Re-invite with SDP do I have > > to repeat also the: > > > > a=inactive > > > > attribute or do I send now my SDP again with > > > > a=sendrecv > > > > > > > > Thanks and Best regards > > > > Joerg > > > > > > > > > _______________________________________________ > > This is the SIP Forum discussion mailing list > > TO UNSUBSCRIBE, or edit your delivery options, > > please visit > > http://sipforum.org/mailman/listinfo/discussion > > Post to the list at discussion at sipforum.org > > > > > > > ____________________________________________________________________________ > ________ > Yahoo! oneSearch: Finally, mobile search > that gives answers, not web links. > http://mobile.yahoo.com/mobileweb/onesearch?refer=1ONXIC > ____________________________________________________________________________________ Check out the hottest 2008 models today at Yahoo! Autos. http://autos.yahoo.com/new_cars.html From victor.pascual.avila at gmail.com Tue Oct 2 08:23:15 2007 From: victor.pascual.avila at gmail.com (=?ISO-8859-1?Q?Victor_Pascual_=C1vila?=) Date: Tue, 2 Oct 2007 14:23:15 +0200 Subject: [SIPForum-discussion] border controller for sip In-Reply-To: <20071001081513.32247.qmail@f5mail-237-201.rediffmail.com> References: <20071001081513.32247.qmail@f5mail-237-201.rediffmail.com> Message-ID: <618e24240710020523m2896f194l9b4ba8c0159f4c54@mail.gmail.com> Hello, check the following link. http://www.opensipstack.org/sbc_man_quickstart.html I hope it'll be useful, Victor Pascual On 1 Oct 2007 08:15:13 -0000, devesh bissa wrote: > > Hi, > I want to use border controller(openSBC) with IMS network for sip. > Please help (how to configure and use it) if anyone did it previously. > > Thank you > Devesh > [image: sig js] > _______________________________________________ > This is the SIP Forum discussion mailing list > TO UNSUBSCRIBE, or edit your delivery options, please visit > http://sipforum.org/mailman/listinfo/discussion > Post to the list at discussion at sipforum.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071002/352b77c2/attachment.html From meharchaoui28 at googlemail.com Tue Oct 2 08:47:58 2007 From: meharchaoui28 at googlemail.com (mohammed El harchaoui) Date: Tue, 2 Oct 2007 14:47:58 +0200 Subject: [SIPForum-discussion] CSDM problem Message-ID: <6f011d50710020547k64813c97ofb8873da7185d7f5@mail.gmail.com> Hi all, I'm trying to develop a sip mobile client, that uses xcap to manage resource list. so if the client needs for example to modify a resource list, he MUST send a "PUT" http message to the csdm server, but the problem is that the mobile devices on which the client should be run do not support the mentioned method above(PUT and also DELETE). DOes anyone solved this problem before, plzzzz help me!!! Mohammed -------------- next part -------------- An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071002/0fc4d2e6/attachment.html From PScheffler at carrieraccess.com Tue Oct 2 09:59:11 2007 From: PScheffler at carrieraccess.com (Scheffler, Paul) Date: Tue, 2 Oct 2007 07:59:11 -0600 Subject: [SIPForum-discussion] specific case with a=inactive In-Reply-To: Message-ID: <33E402324D746F48AF9FBAC491FF5C8C4CE6C1@camailsvr01.carrieraccess.com> Hello Joerg: My recommendation (when you get a re-INVITE with no SDP) is to ignore the current call state, and send an SDP response in the 200 OK which represents your normal offer when initiating a new call. There are some SIP application servers which expect this, because they are trying to determine your normal capabilities in preparation for reconnecting your call. If you do this, you should not have audio problems. Paul Scheffler Carrier Access Corp. Boulder, CO ________________________________ From: discussion-bounces at sipforum.org [mailto:discussion-bounces at sipforum.org] On Behalf Of Joerg Scholz Sent: Tuesday, October 02, 2007 3:05 AM To: 'badal naik'; Traussnig Robert; discussion at sipforum.org Subject: Re: [SIPForum-discussion] specific case with a=inactive Thanks Badel and Robert for your responds. That's also what I expected to do - but it seems to be a problem for that class 5 SIP server solution which I try to work with. It uses the Re-invite without SDP to retrieve the hold call. Thereby the retrieve is initiated with a user agent web application (not the phone). If I send still: A=inactive In the SDP of the 200ok response of my device; it is used for the second leg of the call as SDP content and the call will be reestablished without voice. So I need exactly to know what the right behavior is in that case. It is quite simple to change the behavior but I'm pretty sure that other cases will have problem then. Thanks again and best regards Joerg -----Original Message----- From: badal naik [mailto:kunusan at yahoo.com] Sent: Tuesday, October 02, 2007 10:41 AM To: Traussnig Robert; Joerg Scholz; discussion at sipforum.org Subject: Re: [SIPForum-discussion] specific case with a=inactive Well, I think in your scenario U have to add the SDP parameter of the previous success(2**) reponse. Re-Invite should work as follows in ur case: When a UAC sends a re-invite with no session description, in which case the first reliable non-failure response to the re-invite will contain the offer. That is my understanding.. Thanks Badal Naik --- Traussnig Robert wrote: > Hi! > > I'm not quite sure if this is correct but because of > the fact that you > didn't get a SDP in the Re-Invite you have to give > an offer in your > 200ok. So you have to decide if you send a=inactive > again or a=sendrecv. > Maybe someone have a better answer. > > Best Regards, > Robert > > > -----Ursprungliche Nachricht----- > Von: discussion-bounces at sipforum.org > [mailto:discussion-bounces at sipforum.org]Im Auftrag > von Joerg Scholz > Gesendet: Montag, 01. Oktober 2007 18:58 > An: discussion at sipforum.org > Betreff: [SIPForum-discussion] specific case with > a=inactive > > > > Hello everybody, > > I have a quick question related to the SDP signaling > behavior in a > specific hold case. My scenario is the following: > > > > Incoming Re-invite of an established voice call with > the following > content of SDP: > > c=IN IP4 0.0.0.0 > > ... > > a=inactive > > > > In the 200ok I confirm the attribute line: > > a=inactive > > > > and the call is hold - everything ok. > > Later on comes another Re-invite without SDP for the > same call. Now my > question, if I answer in the 200ok for that > Re-invite with SDP do I have > to repeat also the: > > a=inactive > > attribute or do I send now my SDP again with > > a=sendrecv > > > > Thanks and Best regards > > Joerg > > > > > _______________________________________________ > This is the SIP Forum discussion mailing list > TO UNSUBSCRIBE, or edit your delivery options, > please visit > http://sipforum.org/mailman/listinfo/discussion > Post to the list at discussion at sipforum.org > ________________________________________________________________________ ____________ Yahoo! oneSearch: Finally, mobile search that gives answers, not web links. http://mobile.yahoo.com/mobileweb/onesearch?refer=1ONXIC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071002/fe956741/attachment-0001.html From j.scholz at teles.de Tue Oct 2 10:22:00 2007 From: j.scholz at teles.de (Joerg Scholz) Date: Tue, 2 Oct 2007 16:22:00 +0200 Subject: [SIPForum-discussion] specific case with a=inactive Message-ID: Dear Paul, Thank you; I will do so and ignore for now other optional scenarios which might have a problem with that implementation. As I understand it at the moment is there no clear solution for that. Best regards Joerg -----Original Message----- From: Scheffler, Paul [mailto:PScheffler at carrieraccess.com] Sent: Tuesday, October 02, 2007 3:59 PM To: Joerg Scholz; badal naik; Traussnig Robert; discussion at sipforum.org Subject: RE: [SIPForum-discussion] specific case with a=inactive Hello Joerg: My recommendation (when you get a re-INVITE with no SDP) is to ignore the current call state, and send an SDP response in the 200 OK which represents your normal offer when initiating a new call. There are some SIP application servers which expect this, because they are trying to determine your normal capabilities in preparation for reconnecting your call. If you do this, you should not have audio problems. Paul Scheffler Carrier Access Corp. Boulder, CO _____ From: discussion-bounces at sipforum.org [mailto:discussion-bounces at sipforum.org] On Behalf Of Joerg Scholz Sent: Tuesday, October 02, 2007 3:05 AM To: 'badal naik'; Traussnig Robert; discussion at sipforum.org Subject: Re: [SIPForum-discussion] specific case with a=inactive Thanks Badel and Robert for your responds. That's also what I expected to do - but it seems to be a problem for that class 5 SIP server solution which I try to work with. It uses the Re-invite without SDP to retrieve the hold call. Thereby the retrieve is initiated with a user agent web application (not the phone). If I send still: A=inactive In the SDP of the 200ok response of my device; it is used for the second leg of the call as SDP content and the call will be reestablished without voice. So I need exactly to know what the right behavior is in that case. It is quite simple to change the behavior but I'm pretty sure that other cases will have problem then. Thanks again and best regards Joerg -----Original Message----- From: badal naik [mailto:kunusan at yahoo.com ] Sent: Tuesday, October 02, 2007 10:41 AM To: Traussnig Robert; Joerg Scholz; discussion at sipforum.org Subject: Re: [SIPForum-discussion] specific case with a=inactive Well, I think in your scenario U have to add the SDP parameter of the previous success(2**) reponse. Re-Invite should work as follows in ur case: When a UAC sends a re-invite with no session description, in which case the first reliable non-failure response to the re-invite will contain the offer. That is my understanding.. Thanks Badal Naik --- Traussnig Robert wrote: > Hi! > > I'm not quite sure if this is correct but because of > the fact that you > didn't get a SDP in the Re-Invite you have to give > an offer in your > 200ok. So you have to decide if you send a=inactive > again or a=sendrecv. > Maybe someone have a better answer. > > Best Regards, > Robert > > > -----Ursprungliche Nachricht----- > Von: discussion-bounces at sipforum.org > [mailto:discussion-bounces at sipforum.org ]Im Auftrag > von Joerg Scholz > Gesendet: Montag, 01. Oktober 2007 18:58 > An: discussion at sipforum.org > Betreff: [SIPForum-discussion] specific case with > a=inactive > > > > Hello everybody, > > I have a quick question related to the SDP signaling > behavior in a > specific hold case. My scenario is the following: > > > > Incoming Re-invite of an established voice call with > the following > content of SDP: > > c=IN IP4 0.0.0.0 > > ... > > a=inactive > > > > In the 200ok I confirm the attribute line: > > a=inactive > > > > and the call is hold - everything ok. > > Later on comes another Re-invite without SDP for the > same call. Now my > question, if I answer in the 200ok for that > Re-invite with SDP do I have > to repeat also the: > > a=inactive > > attribute or do I send now my SDP again with > > a=sendrecv > > > > Thanks and Best regards > > Joerg > > > > > _______________________________________________ > This is the SIP Forum discussion mailing list > TO UNSUBSCRIBE, or edit your delivery options, > please visit > http://sipforum.org/mailman/listinfo/discussion > Post to the list at discussion at sipforum.org > ____________________________________________________________________________ ________ Yahoo! oneSearch: Finally, mobile search that gives answers, not web links. http://mobile.yahoo.com/mobileweb/onesearch?refer=1ONXIC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071002/4190ccdb/attachment.html From zmihaly at madein.hu Tue Oct 2 12:07:10 2007 From: zmihaly at madein.hu (Mihaly Zachar) Date: Tue, 02 Oct 2007 18:07:10 +0200 Subject: [SIPForum-discussion] 302 message question Message-ID: <47026CAE.6030603@madein.hu> Hi all, I'm writing an UAS. The UAS has a feature, that if the called number is matching with a pattern, it will send 183 Session in progress, than plays an RTP stream and then redirect the UAC with 302 Moved Temporarily.. There is an UAC, and it's developers says that I should not send 302 Redirect after the 183 Session in progress. This solution works well with CISCO media gateways. I can't find it in the RFC 3261 who has the truth.. Can sy help me in this ? So, the call flow is the following: UAC UAS --- INVITE ---> <--- 100 ------ <--- 183 ------ <-- RTP -- . . . <--- 302 ---- Is this correct ? Thanks, Misi From sreekant_nair at yahoo.com Tue Oct 2 14:05:42 2007 From: sreekant_nair at yahoo.com (sreekant nair) Date: Tue, 2 Oct 2007 11:05:42 -0700 (PDT) Subject: [SIPForum-discussion] 302 message question Message-ID: <11724.69746.qm@web51107.mail.re2.yahoo.com> Check out this link. http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/rel_docs/sip_flo/hennigan.htm There is a call flow depicting the messaging for a scenario where a 3XX response is received after a 183 is sent by the server. I guess that explains how CISCO supports it. But yeah I need to dig deeper to find an RFC that states this. Regards Sreekant ----- Original Message ---- From: Mihaly Zachar To: discussion at sipforum.org Sent: Tuesday, October 2, 2007 12:07:10 PM Subject: [SIPForum-discussion] 302 message question Hi all, I'm writing an UAS. The UAS has a feature, that if the called number is matching with a pattern, it will send 183 Session in progress, than plays an RTP stream and then redirect the UAC with 302 Moved Temporarily.. There is an UAC, and it's developers says that I should not send 302 Redirect after the 183 Session in progress. This solution works well with CISCO media gateways. I can't find it in the RFC 3261 who has the truth.. Can sy help me in this ? So, the call flow is the following: UAC UAS --- INVITE ---> <--- 100 ------ <--- 183 ------ <-- RTP -- . . . <--- 302 ---- Is this correct ? Thanks, Misi _______________________________________________ This is the SIP Forum discussion mailing list TO UNSUBSCRIBE, or edit your delivery options, please visit http://sipforum.org/mailman/listinfo/discussion Post to the list at discussion at sipforum.org ____________________________________________________________________________________ Looking for a deal? Find great prices on flights and hotels with Yahoo! FareChase. http://farechase.yahoo.com/ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071002/84d7eb88/attachment.html From amos.halfon at gmail.com Wed Oct 3 01:49:20 2007 From: amos.halfon at gmail.com (Amos Halfon) Date: Wed, 3 Oct 2007 07:49:20 +0200 Subject: [SIPForum-discussion] (no subject) Message-ID: <697963e10710022249o2bd13e6ayd95a4d6c12a97794@mail.gmail.com> -------------- next part -------------- An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071003/5fddd573/attachment.html From louis at ttv.com.hk Wed Oct 3 02:11:58 2007 From: louis at ttv.com.hk (Louis Wu) Date: Wed, 3 Oct 2007 14:11:58 +0800 Subject: [SIPForum-discussion] Call Disconnect issue with Cisco AS5300 running SIP Message-ID: <20071003055430.M56996@ttv.com.hk> Hi All, I have a Cisco AS5300 using SIP and initiate SIP call to a SIP server YATE (v 1.3.0). I have a call disconnect problem whenever my Cisco receive a 183 Session Progress message from the YATE server. The symptoms are listed as below. 1. Cisco AS5300 send an INVITE to the YATE server 2. YATE returns a 100 Trying message 3. YATE returns a 183 Session Progress 4. YATE returns a 200 OK 5. Two-way-audio starts (start conversation as usual), but at the ISDN side of the Cisco, the call is shown to be "not connected" 6. Cisco sends a ACK 7. Cisco sends a BYE 8. YATE returns a 100 Trying 9. Cisco sends a BYE 10. YATE returns a 200 OK 11. Call disconnect with status message saying "no answer" at the calling party's mobile handset 12. Cisco logs a Disconnect Cause (CC) : 16 (SIP) : 200 If the YATE returns a 180 Session Progress in (3) above, the call will be connected normally and works as usual. Please give me your professional advice and resolution on the above disconnect issue. Cheers Louis From mhiqe at yahoo.com Wed Oct 3 04:30:41 2007 From: mhiqe at yahoo.com (Mhike) Date: Wed, 3 Oct 2007 01:30:41 -0700 (PDT) Subject: [SIPForum-discussion] FMTP / RTPMAP Message-ID: <486570.72622.qm@web50811.mail.re2.yahoo.com> Hi, Can anyone tell me what's the meaning of FMTP and RTPMAP? What are their differences? Thanks. ____________________________________________________________________________________ Don't let your dream ride pass you by. Make it a reality with Yahoo! Autos. http://autos.yahoo.com/index.html -------------- next part -------------- An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071003/1555632a/attachment.html From abhishek.mishra at globallogic.com Wed Oct 3 04:47:18 2007 From: abhishek.mishra at globallogic.com (Abhishek Mishra) Date: Wed, 03 Oct 2007 14:17:18 +0530 Subject: [SIPForum-discussion] FMTP / RTPMAP In-Reply-To: <486570.72622.qm@web50811.mail.re2.yahoo.com> References: <486570.72622.qm@web50811.mail.re2.yahoo.com> Message-ID: <1191401237.2587.6.camel@linux.site> Hi Mhike, Please refer to RFC 2327 and RFC 3264: http://tools.ietf.org/html/rfc2327 Kind Regards, -Abhishek On Wed, 2007-10-03 at 14:00, Mhike wrote: > Hi, > > Can anyone tell me what's the meaning of FMTP and RTPMAP? What are > their differences? > > Thanks. > > > ______________________________________________________________________ > Fussy? Opinionated? Impossible to please? Perfect. Join Yahoo!'s user > panel and lay it on us. > > ______________________________________________________________________ > _______________________________________________ > This is the SIP Forum discussion mailing list > TO UNSUBSCRIBE, or edit your delivery options, please visit http://sipforum.org/mailman/listinfo/discussion > Post to the list at discussion at sipforum.org > From ian.sivell at gmail.com Wed Oct 3 07:36:29 2007 From: ian.sivell at gmail.com (Ian Sivell) Date: Wed, 3 Oct 2007 12:36:29 +0100 Subject: [SIPForum-discussion] CISCO 7940 TFTP Timeout Message-ID: <123aa00e0710030436y6996fd33o577aed42e77b49f@mail.gmail.com> Hi, I hope someone can help me I have recently bought a used 7940 from EBay, initially it booted and I could get into the menus (all be them locked), I found on the cisco site to hold down th # key whilst powering up and then enter 123456789*0# to reset toi factory defaults. Since doing is the phone boots but just stays at the tftp timeout message. On the Cisco site it says that the phone should timeout after 20 seconds and continue to boot correctly after that giving access to the menus mine does not seem to do this. It has been loaded with SCCP (Skinny) as far as I can tell, and the DHCP server on my network is assigning it a DHCP address within a valid subnet but still it times out and does not get any further than the message above. I have reset this several times to n avail Please some one help me Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071003/eafd44f7/attachment.html From yasin at kaplan.net Wed Oct 3 07:46:47 2007 From: yasin at kaplan.net (Yasin KAPLAN) Date: Wed, 3 Oct 2007 14:46:47 +0300 Subject: [SIPForum-discussion] TekSIP Message-ID: <015001c805b3$163e84e0$1a0d3ad4@doruk.com.tr> Hi, I've recently released beta version of TekSIP Registrar & Proxy: http://www.teksip.com/ You feedback is welcomed. Thanks, Yasin KAPLAN -------------- next part -------------- An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071003/20a63309/attachment.html From amit.v at pyronetworks.com Wed Oct 3 09:46:03 2007 From: amit.v at pyronetworks.com (amit) Date: Wed, 03 Oct 2007 19:16:03 +0530 Subject: [SIPForum-discussion] Image with invite msg Message-ID: <1191419164.5119.3.camel@amit> Hi All, How we send image with sip invite message ? Thanks in Advance Amit From mohamed2005777 at yahoo.com Wed Oct 3 10:41:03 2007 From: mohamed2005777 at yahoo.com (mohamed hamdy) Date: Wed, 3 Oct 2007 07:41:03 -0700 (PDT) Subject: [SIPForum-discussion] Fwd: Call Disconnect issue with Cisco AS5300 running SIP Message-ID: <963352.71849.qm@web56404.mail.re3.yahoo.com> Hi,every one I'm now working for voip call signalling over sip and i want really a support to download cisco sip proxy server for free ( without paying) plz help me as soon as possible becouse the end of the project will be within month Note: forwarded message attached. --------------------------------- Building a website is a piece of cake. Yahoo! Small Business gives you all the tools to get online. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071003/2306a42f/attachment.html -------------- next part -------------- An embedded message was scrubbed... From: "Louis Wu" Subject: [SIPForum-discussion] Call Disconnect issue with Cisco AS5300 running SIP Date: Wed, 3 Oct 2007 14:11:58 +0800 Size: 3918 Url: http://sipforum.org/pipermail/discussion/attachments/20071003/2306a42f/attachment.mht From raymond.jender.ctr at disa.mil Wed Oct 3 11:28:10 2007 From: raymond.jender.ctr at disa.mil (Jender, Raymond C CTR DISA JITC) Date: Wed, 3 Oct 2007 08:28:10 -0700 Subject: [SIPForum-discussion] Fwd: Call Disconnect issue with Cisco AS5300running SIP (UNCLASSIFIED) In-Reply-To: <963352.71849.qm@web56404.mail.re3.yahoo.com> References: <963352.71849.qm@web56404.mail.re3.yahoo.com> Message-ID: <00E9117C300386479B7E2F3CDBF798F601E7EC5F@pothia.disanet.disa-u.mil> Classification: UNCLASSIFIED Caveats: NONE If the Cisco SIP Proxy Server is a commercial product, you should buy it and not look for bootlegged copies. Otherwie, there are other free sip proxies out there.... Raymond C. Jender Booz|Allen|Hamilton DSN IA Test Team Ft. Huachuca, Az. 520-538-2588 -----Original Message----- From: discussion-bounces at sipforum.org [mailto:discussion-bounces at sipforum.org] On Behalf Of mohamed hamdy Sent: Wednesday, October 03, 2007 7:41 AM To: discussion at sipforum.org Subject: [SIPForum-discussion] Fwd: Call Disconnect issue with Cisco AS5300running SIP Hi,every one I'm now working for voip call signalling over sip and i want really a support to download cisco sip proxy server for free ( without paying) plz help me as soon as possible becouse the end of the project will be within month Note: forwarded message attached. ________________________________ Building a website is a piece of cake. Yahoo! Small Business gives you all the tools to get online. Classification: UNCLASSIFIED Caveats: NONE From gpaul at aylus.com Wed Oct 3 11:43:11 2007 From: gpaul at aylus.com (Geo Paul) Date: Wed, 3 Oct 2007 11:43:11 -0400 Subject: [SIPForum-discussion] FMTP / RTPMAP In-Reply-To: <486570.72622.qm@web50811.mail.re2.yahoo.com> Message-ID: When ever a dynamic payload type is used in the sdp or when ever additional information is required to decode, the additional information should be given in a=rtpmap: /[/] a=fmtp: This attribute allows parameters that are specific to a particular format to be conveyed in a way that SDP doesn't have to understand them. The format must be one of the formats specified for the media. Format-specific parameters may be any set of parameters required to be conveyed by SDP and given unchanged to the media tool that will use this format. It is a media attribute, and is not dependent on charset. Geo _____ From: Mhike [mailto:mhiqe at yahoo.com] Sent: Wednesday, October 03, 2007 4:31 AM To: discussion at sipforum.org Subject: [SIPForum-discussion] FMTP / RTPMAP Hi, Can anyone tell me what's the meaning of FMTP and RTPMAP? What are their differences? Thanks. _____ Fussy? Opinionated? Impossible to please? Perfect. Join Yahoo!'s user panel and lay it on us. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071003/fdab4e89/attachment.html From cross at gocross.com Wed Oct 3 12:30:26 2007 From: cross at gocross.com (Tom Cross) Date: Wed, 3 Oct 2007 10:30:26 -0600 Subject: [SIPForum-discussion] Book Review of "Securing VoIP Networks" Message-ID: <014e01c805da$b6bedbf0$679c0818@dv5020us> Please pass along. "Once in a blue moon you read a book that not just meets but beats your expectations. The book I am referring to is Securing VoIP Networks by Peter Thermos and Ari Takanen, Addison-Wesley, ISBN-0-321-43734-9, www.awprofessional.com . The more you read, the more you want to read. All too often technical books are too-deep or too-high. This book provides practical, understandable and most importantly, implementable (new word) information. As a SIP course developer and trainer, I am always looking for something to help students learn and do more. This book does that and more. Guess by now you know how I feel, so I will stop." Tom Cross - CEO TECHtionary.com Cheers, Join the CrossTalk blog on TMCnet - http://blog.tmcnet.com/cross-talk/ See all the new Digital Communications, VoIP, SIP and Advanced Network courses at: http://www.techtionary.com - TECHtionary - The World's Largest Animated Library on TECHnology Web Hosting Magazine's Editor's Choice for Technical Help -------------- next part -------------- An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071003/076b6004/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/jpeg Size: 2743 bytes Desc: not available Url : http://sipforum.org/pipermail/discussion/attachments/20071003/076b6004/attachment.jpe From chauhan_delhi at yahoo.com Wed Oct 3 12:42:57 2007 From: chauhan_delhi at yahoo.com (Ramesh Chauhan) Date: Wed, 3 Oct 2007 09:42:57 -0700 (PDT) Subject: [SIPForum-discussion] DTMF Issue - Asterisk Message-ID: <309731.93330.qm@web34409.mail.mud.yahoo.com> Hi, I am using Dax IP Phone (Model: DX-301P, H/W Ver:5.1). SIP.CONF : [7711] type=friend username=7711 secret=7711 host=dynamic port=5060 dtmfmode=inband canreinvite=no context=outgoing_ISD mailbox=7711 at default disallow=all allow=ulaw Problem: 7711 is assigned to DAX IP Phone. DTMF is not working. i have tried with already the above settings in SIP.CONF with info, inband, rfc2833, auto. Suggest what to do, so that our DTMF works with this device. Regards Chauhan --------------------------------- Building a website is a piece of cake. Yahoo! Small Business gives you all the tools to get online. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071003/5a5521d4/attachment.html From joel.silva at novabase.pt Wed Oct 3 12:49:43 2007 From: joel.silva at novabase.pt (Joel Silva) Date: Wed, 3 Oct 2007 17:49:43 +0100 Subject: [SIPForum-discussion] Presence for multiple publicIDs References: Message-ID: Hello. I was trying to make an application that give me the presence of a number of pubIds associate with a privateId. Imagine that I have associated with my privateId, two publicIDs, one for my sip phone and other to my sip app. I would like that other users could see my name and then a tree associated with the presence of my publicIds. Is this possible? How can I do it? Thanks, Joel. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071003/885deb94/attachment.html From victor.pascual.avila at gmail.com Wed Oct 3 13:34:34 2007 From: victor.pascual.avila at gmail.com (=?ISO-8859-1?Q?Victor_Pascual_=C1vila?=) Date: Wed, 3 Oct 2007 19:34:34 +0200 Subject: [SIPForum-discussion] TekSIP In-Reply-To: <015001c805b3$163e84e0$1a0d3ad4@doruk.com.tr> References: <015001c805b3$163e84e0$1a0d3ad4@doruk.com.tr> Message-ID: <618e24240710031034ic29d72byabfceaf42480f357@mail.gmail.com> Hello Yasin, good job. Have you tested it with several users (stress test) ? Regards, Victor Pascual On 03/10/2007, Yasin KAPLAN wrote: > > > > > Hi, > > > > I've recently released beta version of TekSIP Registrar & Proxy: > > > > http://www.teksip.com/ > > > > You feedback is welcomed. > > > > Thanks, > > > > Yasin KAPLAN > > > > > _______________________________________________ > This is the SIP Forum discussion mailing list > TO UNSUBSCRIBE, or edit your delivery options, please visit > http://sipforum.org/mailman/listinfo/discussion > Post to the list at discussion at sipforum.org > > From lingyunxjtu at gmail.com Thu Oct 4 02:42:27 2007 From: lingyunxjtu at gmail.com (Karl Tian) Date: Thu, 4 Oct 2007 14:42:27 +0800 Subject: [SIPForum-discussion] The max duration of SIP conversation Message-ID: <807efb400710032342rdb8ca9fyc456c7a3ab622060@mail.gmail.com> Hello everyone, Who can tell me if some rfc protocal(for example: rfc3261) has define the max duration of a sip conversation as 72 hours? Now I'm testing the haleness for a kind of sip client, but the conversations of those clients all stop when the duration reachs to 72 hours. I guess that the question may be caused by a special timer. Please help me about this, thanks! -- Karl.Tian Infinite Shanghai Communication Terminals Ltd. Email :lingyunxjtu at gmail.com Msn:lingyunxjtu at hotmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071004/125c1b4d/attachment.html From VPFR47 at motorola.com Thu Oct 4 02:56:32 2007 From: VPFR47 at motorola.com (S Selvakumar-VPFR47) Date: Thu, 4 Oct 2007 14:56:32 +0800 Subject: [SIPForum-discussion] Asterisk Usage Message-ID: <40E89886C8B3B54B98C5291646C591AA01B6E54A@ZMY16EXM67.ds.mot.com> Hi, I want to know some info about Asterisk. Can anyone help me regards Selva -------------- next part -------------- An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071004/1c7d274c/attachment-0001.html From bn.darshan at gmail.com Thu Oct 4 03:54:36 2007 From: bn.darshan at gmail.com (darshan b n) Date: Thu, 4 Oct 2007 13:24:36 +0530 Subject: [SIPForum-discussion] sips message in ethereal Message-ID: <555d2a660710040054i231a803bua71e70caf2a731ae@mail.gmail.com> Hi all, I want know how to capture a sips message in ethereal? If not is there any other free network protocol analyzer which support this feature? Reply ASAP -- Darshan B N Thanks & Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071004/d9bd6774/attachment.html From kunusan at yahoo.com Thu Oct 4 04:13:50 2007 From: kunusan at yahoo.com (badal naik) Date: Thu, 4 Oct 2007 01:13:50 -0700 (PDT) Subject: [SIPForum-discussion] The max duration of SIP conversation In-Reply-To: <807efb400710032342rdb8ca9fyc456c7a3ab622060@mail.gmail.com> Message-ID: <200842.58269.qm@web56703.mail.re3.yahoo.com> Karl, as i know there is no special timer that controls a SIP Session.I have the experience of media session open over 100 hours. One thing i can suggest u, Please check your session by session timing from ethereal capture. Check how much time each packet is taking for round trip, how much is jitter value etc etc. May be u can get a clue from that. Unless and until I see the packet capture I can't guess the reason.It is purely your environment and setting issue.Nothing related to SIP universal implementation. To get the details of timer used in SIP, U can go to RFC3261 and seach there. To make ur effort more easy search"Summary of timers". Thanks Badal Naik --- Karl Tian wrote: > Hello everyone, > Who can tell me if some rfc protocal(for > example: rfc3261) has define > the max duration of a sip conversation as 72 hours? > Now I'm testing the haleness for a kind of sip > client, but the > conversations of those clients all stop when the > duration reachs to 72 > hours. I guess that the question may be caused by a > special timer. > Please help me about this, thanks! > > > > > -- > Karl.Tian > Infinite Shanghai Communication Terminals Ltd. > Email :lingyunxjtu at gmail.com > Msn:lingyunxjtu at hotmail.com > > _______________________________________________ > This is the SIP Forum discussion mailing list > TO UNSUBSCRIBE, or edit your delivery options, > please visit > http://sipforum.org/mailman/listinfo/discussion > Post to the list at discussion at sipforum.org > ____________________________________________________________________________________ Shape Yahoo! in your own image. Join our Network Research Panel today! http://surveylink.yahoo.com/gmrs/yahoo_panel_invite.asp?a=7 From bn.darshan at gmail.com Thu Oct 4 04:26:23 2007 From: bn.darshan at gmail.com (darshan b n) Date: Thu, 4 Oct 2007 13:56:23 +0530 Subject: [SIPForum-discussion] sips message in ethereal In-Reply-To: <345812.11216.qm@web8320.mail.in.yahoo.com> References: <555d2a660710040054i231a803bua71e70caf2a731ae@mail.gmail.com> <345812.11216.qm@web8320.mail.in.yahoo.com> Message-ID: <555d2a660710040126l3c3bf225qab0b2e1872165607@mail.gmail.com> why it is not able to capture yahoo messages which runs on sip ? On 04/10/2007, Lakshminarayanan.Ramasami wrote: > > type in filter textbox like this sip > > *darshan b n * wrote: > > > Hi all, > > I want know how to capture a sips message in ethereal? > > If not is there any other free network protocol analyzer which support > this feature? > > Reply ASAP > > -- > Darshan B N > > Thanks & Regards > _______________________________________________ > This is the SIP Forum discussion mailing list > TO UNSUBSCRIBE, or edit your delivery options, please visit > http://sipforum.org/mailman/listinfo/discussion > Post to the list at discussion at sipforum.org > > > > > Thanks and Regards, > Lakshminarayanan.Ramasami > Cell : +91-9980840622,+91-80-41227491 > (BANGALORE) Cell : +91-9840264214,+91-4153-252636 (THIRUKKOYILUR) > > > ------------------------------ > Bring your gang together - do your thing. Start your group. > > -- Darshan B N Thanks & Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071004/f69027c0/attachment.html From kunusan at yahoo.com Thu Oct 4 04:54:46 2007 From: kunusan at yahoo.com (badal naik) Date: Thu, 4 Oct 2007 01:54:46 -0700 (PDT) Subject: [SIPForum-discussion] sips message in ethereal In-Reply-To: <555d2a660710040054i231a803bua71e70caf2a731ae@mail.gmail.com> Message-ID: <864190.51307.qm@web56711.mail.re3.yahoo.com> If you have ethereal Installed in your system, u can follow below instructions, if dont know how to capture the packet. 1.Open The ethereal. 2.Click capture from the main menu. 3.Click Options. 4.In Options, Set the interface upon which you want to receive the packets.Basically it will be an ethernet card. 5. Then Initiate ur SIP activity.And see the contents in Ethereal Tool. 6. In ethereal, there will be one filter.Type there SIP and click Apply. Save this as .cab file if u want to save the file. For more details u can follow ethereal guidelines available at their URL. Others: Analyzer for win32. Sniffer(Sniffer technology) Etherpeek(From Wildpackets) Have not used any of them except ethereal. May be if u wish can do an R&D. Thanks Badal Naik --- darshan b n wrote: > Hi all, > > I want know how to capture a sips message in > ethereal? > > If not is there any other free network protocol > analyzer which support this > feature? > > Reply ASAP > > -- > Darshan B N > > Thanks & Regards > > _______________________________________________ > This is the SIP Forum discussion mailing list > TO UNSUBSCRIBE, or edit your delivery options, > please visit > http://sipforum.org/mailman/listinfo/discussion > Post to the list at discussion at sipforum.org > ____________________________________________________________________________________ Yahoo! oneSearch: Finally, mobile search that gives answers, not web links. http://mobile.yahoo.com/mobileweb/onesearch?refer=1ONXIC From victor.pascual.avila at gmail.com Thu Oct 4 05:09:56 2007 From: victor.pascual.avila at gmail.com (=?ISO-8859-1?Q?Victor_Pascual_=C1vila?=) Date: Thu, 4 Oct 2007 11:09:56 +0200 Subject: [SIPForum-discussion] sips message in ethereal In-Reply-To: <555d2a660710040126l3c3bf225qab0b2e1872165607@mail.gmail.com> References: <555d2a660710040054i231a803bua71e70caf2a731ae@mail.gmail.com> <345812.11216.qm@web8320.mail.in.yahoo.com> <555d2a660710040126l3c3bf225qab0b2e1872165607@mail.gmail.com> Message-ID: <618e24240710040209q6d3d487eh3d6b934272c27971@mail.gmail.com> Hi, use the filter 'sip' or capture by used port. Regards, Victor Pascual -------------- next part -------------- An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071004/08940a62/attachment.html From shakthi_msc at yahoo.com Thu Oct 4 05:20:16 2007 From: shakthi_msc at yahoo.com (shakthi_msc) Date: 4 Oct 2007 02:20:16 -0700 Subject: [SIPForum-discussion] Do we like the same books? Message-ID: <200710040920.l949KI4L011827@sipforum.org> I just joined Shelfari to connect with other book lovers. Come see the books I love and see if we have any in common. Then pick my next book so I can keep on reading. Click below to join my group of friends on Shelfari! http://www.shelfari.com/Register.aspx?ActivityId=22465602&InvitationCode=b46e3bcf-4169-42b2-bb6d-a9515f7ea3b5 shakthi_msc Shelfari is a free site that lets you share book ratings and reviews with friends and meet people who have similar tastes in books. It also lets you build an online bookshelf, join book clubs, and get good book recommendations from friends. You should check it out. -------- You have received this email because shakthi_msc (shakthi_msc at yahoo.com) directly invited you to join his/her community on Shelfari. It is against Shelfari's policies to invite people who you don't know directly. Follow this link (http://www.shelfari.com/actions/emailoptout.aspx?email=discussion at sipforum.org&activityid=22465602) to prevent future invitations to this address. If you believe you do not know this person, you may view (http://www.shelfari.com/shakthi_msc) his/her Shelfari page or report him/her in our feedback (http://www.shelfari.com/Feedback.aspx) section. Shelfari, 616 1st Ave #300, Seattle, WA 98104 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071004/2d8a2026/attachment.html From radhashyambehera at gmail.com Thu Oct 4 06:38:35 2007 From: radhashyambehera at gmail.com (Radhashyam Behera) Date: Thu, 4 Oct 2007 16:08:35 +0530 Subject: [SIPForum-discussion] Requirement in RTP & IMS Team Message-ID: <7c4722c20710040338o2b385d87xc4d1f474f789d558@mail.gmail.com> Hi This is Radhashyam Behera, working with NetHawk Network INdia Pvt Ltd, India. We have urgent requirement in RTP and IMS Team. Interested candidate can apply with the same subject line to me. Detail requirement is given below: *RTP Team:* * * *Resource Requirement: 2* The candidate must have the below criteria. 1. 3+ years of experience. 2. Good C/C++ programming skill (application layer programming experience). 3. Socket programming, 4. Multithreaded programming experience. 5. Familiar with Linux environment. 6. Good knowledge of Real Time protocol, TCP/IP stacks. 7. Good Kernel programming skill and have experience in designing (module level designing experience will be enough) networking protocol state machines. * * *IMS Team:* * * *Resource Requirement: 2 * - Experience ? at least 2+yrs - Basic concepts of Protocols like Diameter, MEGACO, DNS, DHCP, XCAP, SOAP, HTTP, CAMEL, MGCP. - SIP protocol knowledge. - Good Knowledge in IMS Architecture with Interface idea. - Good communication skill - Good Documentation & Presentation. If the candidate knows the below tools, it will be added advantage: - Strong EAST Tool knowledge - Quality Center. - M5 and Ethereal Analyzer tool knowledge. With Regards, Radhashyam -------------- next part -------------- An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071004/ff70e87e/attachment.html From joel.silva at novabase.pt Thu Oct 4 07:31:56 2007 From: joel.silva at novabase.pt (Joel Silva) Date: Thu, 4 Oct 2007 12:31:56 +0100 Subject: [SIPForum-discussion] Registar a client Java References: <555d2a660710040054i231a803bua71e70caf2a731ae@mail.gmail.com><345812.11216.qm@web8320.mail.in.yahoo.com><555d2a660710040126l3c3bf225qab0b2e1872165607@mail.gmail.com> <618e24240710040209q6d3d487eh3d6b934272c27971@mail.gmail.com> Message-ID: Does anybody have a simple example in Java of a client registering in the registrar server? I?m only interested in the part of the client. Thanks, Joel. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071004/9a97e347/attachment-0001.html From chahn at mytelepath.com Thu Oct 4 09:45:09 2007 From: chahn at mytelepath.com (Chris Hahn) Date: Thu, 04 Oct 2007 08:45:09 -0500 Subject: [SIPForum-discussion] Registar a client Java In-Reply-To: Message-ID: <731ec9f5e63a5844a889ca4565122ffc@mytelepath.com> Can I please be removed from this list? I've made several requests via the SIPForum website. Thanks, _____ From: discussion-bounces at sipforum.org [mailto:discussion-bounces at sipforum.org] On Behalf Of Joel Silva Sent: Thursday, October 04, 2007 6:32 AM To: discussion at sipforum.org Subject: [SIPForum-discussion] Registar a client Java Does anybody have a simple example in Java of a client registering in the registrar server? I?m only interested in the part of the client. Thanks, Joel. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071004/b8165db8/attachment.html From raymond.jender.ctr at disa.mil Thu Oct 4 10:37:27 2007 From: raymond.jender.ctr at disa.mil (Jender, Raymond C CTR DISA JITC) Date: Thu, 4 Oct 2007 07:37:27 -0700 Subject: [SIPForum-discussion] Asterisk Usage (UNCLASSIFIED) In-Reply-To: <40E89886C8B3B54B98C5291646C591AA01B6E54A@ZMY16EXM67.ds.mot.com> References: <40E89886C8B3B54B98C5291646C591AA01B6E54A@ZMY16EXM67.ds.mot.com> Message-ID: <00E9117C300386479B7E2F3CDBF798F601E7EC62@pothia.disanet.disa-u.mil> Classification: UNCLASSIFIED Caveats: NONE Go to http://www.asterisk.org Raymond C. Jender Booz|Allen|Hamilton DSN IA Test Team Ft. Huachuca, Az. 520-538-2588 -----Original Message----- From: discussion-bounces at sipforum.org [mailto:discussion-bounces at sipforum.org] On Behalf Of S Selvakumar-VPFR47 Sent: Wednesday, October 03, 2007 11:57 PM To: discussion at sipforum.org Subject: [SIPForum-discussion] Asterisk Usage Hi, I want to know some info about Asterisk. Can anyone help me regards Selva Classification: UNCLASSIFIED Caveats: NONE From abdel_mameri at hotmail.com Thu Oct 4 10:55:18 2007 From: abdel_mameri at hotmail.com (mameri abdelhamid) Date: Thu, 04 Oct 2007 14:55:18 +0000 Subject: [SIPForum-discussion] Can I please be removed from this list? In-Reply-To: <731ec9f5e63a5844a889ca4565122ffc@mytelepath.com> Message-ID: An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071004/abb6ea00/attachment.html From bn.darshan at gmail.com Fri Oct 5 03:47:05 2007 From: bn.darshan at gmail.com (darshan b n) Date: Fri, 5 Oct 2007 13:17:05 +0530 Subject: [SIPForum-discussion] Rtp testing tool Message-ID: <555d2a660710050047t308cd98dicbf4462c70a68901@mail.gmail.com> Hi all, I want to test RTP protocol (g.711,g.729..........AMR)..........Can you please suggest open souce tool which gives statistics such as MOS ,JITTER,LATENCY....etc -- Darshan B N Thanks & Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071005/5dd12986/attachment.html From atutush at yahoo.com Fri Oct 5 04:55:16 2007 From: atutush at yahoo.com (ahmet tutus) Date: Fri, 5 Oct 2007 11:55:16 +0300 (EEST) Subject: [SIPForum-discussion] ACK problem Message-ID: <268147.96922.qm@web53303.mail.re2.yahoo.com> Hi all. I send ACK from Client1 to Client2, and Client 2 can not receive the ACK. When I look at the logs, I see that there are 4 ACKs in the system, but there is no tag in the To header part of the last ACK. I use Route:[routes] keyword and "rrs=true" in the last received OK part. I take also some INFO messages. Why do I take it? Any error in these codes? Any idea about this situation? Thanks in advance, Best Regards... Ahmet TUTUS TURKEY Bogazici University System Control Engineering --------------------------------- Yahoo! kullaniyor musunuz? Istenmeyen postadan biktiniz mi? Istenmeyen postadan en iyi korunma Yahoo! Posta'da http://tr.mail.yahoo.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071005/77a5ebee/attachment.html From somi.suresh at gmail.com Fri Oct 5 05:41:33 2007 From: somi.suresh at gmail.com (cheetah) Date: Fri, 5 Oct 2007 15:11:33 +0530 Subject: [SIPForum-discussion] AMR codec required for the softphone Message-ID: <86f9a8d10710050241x7fea04v1e944c5fd4eba76d@mail.gmail.com> Hi, I need softphone which enabled with AMR Codec. If anybody has details reply to me ASAP From smanickam at velankani.com Fri Oct 5 06:07:40 2007 From: smanickam at velankani.com (Shankar Manickam) Date: Fri, 5 Oct 2007 15:37:40 +0530 Subject: [SIPForum-discussion] CSeq in re-Invite Message-ID: <010d01c80737$906edf00$0e1c000a@blr.velankani.com> Hi, Is there any situation where cseq will get incremented for re-Invite? With Regards, Shankar Manickam -------------- next part -------------- An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071005/719c7831/attachment.html From radhashyambehera at gmail.com Fri Oct 5 06:45:10 2007 From: radhashyambehera at gmail.com (Radhashyam Behera) Date: Fri, 5 Oct 2007 16:15:10 +0530 Subject: [SIPForum-discussion] CSeq in re-Invite In-Reply-To: <010d01c80737$906edf00$0e1c000a@blr.velankani.com> References: <010d01c80737$906edf00$0e1c000a@blr.velankani.com> Message-ID: <7c4722c20710050345h3feac90xdf09b381614ffcc2@mail.gmail.com> In case of re-Invite CSeq needs to be incremented always for the same Dialog. Thanks & Regards, Radhashyam On 10/5/07, Shankar Manickam wrote: > > Hi, > > Is there any situation where cseq will get incremented for re-Invite? > > > > > > *With Regards,* > > *Shankar Manickam* > > > > _______________________________________________ > This is the SIP Forum discussion mailing list > TO UNSUBSCRIBE, or edit your delivery options, please visit > http://sipforum.org/mailman/listinfo/discussion > Post to the list at discussion at sipforum.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071005/b1f67f96/attachment.html From bn.darshan at gmail.com Fri Oct 5 07:16:17 2007 From: bn.darshan at gmail.com (darshan b n) Date: Fri, 5 Oct 2007 16:46:17 +0530 Subject: [SIPForum-discussion] CSeq in re-Invite In-Reply-To: <7c4722c20710050345h3feac90xdf09b381614ffcc2@mail.gmail.com> References: <010d01c80737$906edf00$0e1c000a@blr.velankani.com> <7c4722c20710050345h3feac90xdf09b381614ffcc2@mail.gmail.com> Message-ID: <555d2a660710050416g2c9af425l50300597c32190f5@mail.gmail.com> Call hold feature where u can find re-invite with cseq : incremented Regards darshan b n On 05/10/2007, Radhashyam Behera wrote: > > In case of re-Invite CSeq needs to be incremented always for the same > Dialog. > > Thanks & Regards, > Radhashyam > > On 10/5/07, Shankar Manickam wrote: > > > Hi, > > > > Is there any situation where cseq will get incremented for > > re-Invite? > > > > > > > > > > > > *With Regards,* > > > > *Shankar Manickam* > > > > > > > > _______________________________________________ > > This is the SIP Forum discussion mailing list > > TO UNSUBSCRIBE, or edit your delivery options, please visit > > http://sipforum.org/mailman/listinfo/discussion > > Post to the list at discussion at sipforum.org > > > > > > _______________________________________________ > This is the SIP Forum discussion mailing list > TO UNSUBSCRIBE, or edit your delivery options, please visit > http://sipforum.org/mailman/listinfo/discussion > Post to the list at discussion at sipforum.org > > -- Darshan B N Thanks & Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071005/0dcf5cc3/attachment.html From shiv.kumr at gmail.com Fri Oct 5 07:51:25 2007 From: shiv.kumr at gmail.com (Shiv) Date: Fri, 5 Oct 2007 17:21:25 +0530 Subject: [SIPForum-discussion] CSeq in re-Invite In-Reply-To: <555d2a660710050416g2c9af425l50300597c32190f5@mail.gmail.com> References: <010d01c80737$906edf00$0e1c000a@blr.velankani.com> <7c4722c20710050345h3feac90xdf09b381614ffcc2@mail.gmail.com> <555d2a660710050416g2c9af425l50300597c32190f5@mail.gmail.com> Message-ID: <2a6a20ec0710050451q2ab42047o9e368a83b4bfe841@mail.gmail.com> I suppose, three re-invite will be involved for the same dialog on scenario like : a. hold b. conecting to Music c. connecting back to the endpoint, Since it is same dialog, Cseq will be incremented. On 10/5/07, darshan b n wrote: > > Call hold feature where u can find re-invite with cseq : incremented > > Regards > darshan b n > > > On 05/10/2007, Radhashyam Behera wrote: > > > > In case of re-Invite CSeq needs to be incremented always for the same > > Dialog. > > > > Thanks & Regards, > > Radhashyam > > > > On 10/5/07, Shankar Manickam < smanickam at velankani.com> wrote: > > > > > Hi, > > > > > > Is there any situation where cseq will get incremented for > > > re-Invite? > > > > > > > > > > > > > > > > > > *With Regards,* > > > > > > *Shankar Manickam* > > > > > > > > > > > > _______________________________________________ > > > This is the SIP Forum discussion mailing list > > > TO UNSUBSCRIBE, or edit your delivery options, please visit > > > http://sipforum.org/mailman/listinfo/discussion > > > Post to the list at discussion at sipforum.org > > > > > > > > > > _______________________________________________ > > This is the SIP Forum discussion mailing list > > TO UNSUBSCRIBE, or edit your delivery options, please visit > > http://sipforum.org/mailman/listinfo/discussion > > Post to the list at discussion at sipforum.org > > > > > > > -- > Darshan B N > > Thanks & Regards > _______________________________________________ > This is the SIP Forum discussion mailing list > TO UNSUBSCRIBE, or edit your delivery options, please visit > http://sipforum.org/mailman/listinfo/discussion > Post to the list at discussion at sipforum.org > > -- With best regards, Sivakumar Arumugam +91-9282419188 Alcatel-Lucent. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071005/059a2d94/attachment-0001.html From VPFR47 at motorola.com Fri Oct 5 11:33:26 2007 From: VPFR47 at motorola.com (S Selvakumar-VPFR47) Date: Fri, 5 Oct 2007 23:33:26 +0800 Subject: [SIPForum-discussion] CSeq in re-Invite In-Reply-To: <010d01c80737$906edf00$0e1c000a@blr.velankani.com> Message-ID: <40E89886C8B3B54B98C5291646C591AA01BA21FA@ZMY16EXM67.ds.mot.com> Hi Shankar, Try a Call Hold scenario, where you can find Cseq gets incremented for Hold Invite and UnHold Invite regards Selva ________________________________ From: discussion-bounces at sipforum.org [mailto:discussion-bounces at sipforum.org] On Behalf Of Shankar Manickam Sent: Friday, October 05, 2007 3:38 PM To: discussion at sipforum.org Subject: [SIPForum-discussion] CSeq in re-Invite Hi, Is there any situation where cseq will get incremented for re-Invite? With Regards, Shankar Manickam -------------- next part -------------- An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071005/ad96f27f/attachment.html From losmacocos_1 at hotmail.com Sat Oct 6 04:24:43 2007 From: losmacocos_1 at hotmail.com (Ignacio Macocos) Date: Sat, 6 Oct 2007 08:24:43 +0000 Subject: [SIPForum-discussion] Can I please be removed from this list? In-Reply-To: References: <731ec9f5e63a5844a889ca4565122ffc@mytelepath.com> Message-ID: Please me too!!!!! I'dont like to receive mails from this forum. thks From: abdel_mameri at hotmail.comTo: chahn at mytelepath.com; discussion at sipforum.orgDate: Thu, 4 Oct 2007 14:55:18 +0000Subject: [SIPForum-discussion] Can I please be removed from this list? Can I please be removed from this list? I?ve made several requests via the SIPForum website. Thanks, From: "Chris Hahn" To: "discussion at sipforum.org" Subject: Re: [SIPForum-discussion] Registar a client JavaDate: Thu, 04 Oct 2007 08:45:09 -0500 Can I please be removed from this list? I?ve made several requests via the SIPForum website. Thanks, From: discussion-bounces at sipforum.org [mailto:discussion-bounces at sipforum.org] On Behalf Of Joel SilvaSent: Thursday, October 04, 2007 6:32 AMTo: discussion at sipforum.orgSubject: [SIPForum-discussion] Registar a client Java Does anybody have a simple example in Java of a client registering in the registrar server? I?m only interested in the part of the client. Thanks, Joel. >_______________________________________________>This is the SIP Forum discussion mailing list>TO UNSUBSCRIBE, or edit your delivery options, please visit http://sipforum.org/mailman/listinfo/discussion>Post to the list at discussion at sipforum.org Saviez-vous que Windows Live Messenger est disponible d?s maintenant sur votre GSM ? _________________________________________________________________ Climb to the top of the charts!? Play Star Shuffle:? the word scramble challenge with star power. http://club.live.com/star_shuffle.aspx?icid=starshuffle_wlmailtextlink_oct -------------- next part -------------- An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071006/e5ccee2d/attachment.html From amit.v at pyronetworks.com Sat Oct 6 09:08:06 2007 From: amit.v at pyronetworks.com (amit) Date: Sat, 06 Oct 2007 18:38:06 +0530 Subject: [SIPForum-discussion] images with sip messages Message-ID: <1191676086.6009.18.camel@amit> Hi All, Can we send images with sip messages ? if yes, then how it is possible ? Thanks in Advance Amit -------------- next part -------------- An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071006/d22f96f7/attachment.html From amit.v at pyronetworks.com Sat Oct 6 09:10:56 2007 From: amit.v at pyronetworks.com (amit) Date: Sat, 06 Oct 2007 18:40:56 +0530 Subject: [SIPForum-discussion] confrence in sip Message-ID: <1191676256.6009.21.camel@amit> Hi All, I am working on Sip Chat Sever. Sip can support chat conference?????? Thanks in Advance Amit From eshwarry at gmail.com Sat Oct 6 23:02:00 2007 From: eshwarry at gmail.com (eswari s) Date: Sun, 7 Oct 2007 08:32:00 +0530 Subject: [SIPForum-discussion] Which RFC ? Message-ID: <83ea91a60710062002q6f0a9ceev91587bb8a693404b@mail.gmail.com> Hi Which RFC is followed for file transfer in sip phone's ?? best wishes, eshwary -------------- next part -------------- An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071007/0715465a/attachment.html From triveni.prabhu at wipro.com Sun Oct 7 00:49:59 2007 From: triveni.prabhu at wipro.com (triveni.prabhu at wipro.com) Date: Sun, 7 Oct 2007 10:19:59 +0530 Subject: [SIPForum-discussion] Which RFC ? References: <83ea91a60710062002q6f0a9ceev91587bb8a693404b@mail.gmail.com> Message-ID: <4EB051147731D64B9B9187139BAC3E9502ACDE@BLR-SJP-MBX01.wipro.com> Hi, You can refer to http://tools.ietf.org/id/draft-isomaki-sipping-file-transfer-00.txt Thank you, Regards, Triveni Prabhu. ________________________________ From: discussion-bounces at sipforum.org on behalf of eswari s Sent: Sun 10/7/2007 8:32 AM To: discussion at sipforum.org Subject: [SIPForum-discussion] Which RFC ? Hi Which RFC is followed for file transfer in sip phone's ?? best wishes, eshwary -------------- next part -------------- An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071007/2ff501eb/attachment.html From harini.dhanasekaran at wipro.com Sun Oct 7 01:49:56 2007 From: harini.dhanasekaran at wipro.com (harini.dhanasekaran at wipro.com) Date: Sun, 7 Oct 2007 11:19:56 +0530 Subject: [SIPForum-discussion] confrence in sip References: <1191676256.6009.21.camel@amit> Message-ID: Hi Amit, Yes, SIP can support chat or IM conference. Refer to this link: http://www.ietf.org/internet-drafts/draft-ietf-simple-chat-00.txt Note: The link above is an internet draft document valid till dec 14, 2007. Thank you, Regards, Harini Dhanasekaran. ________________________________ From: discussion-bounces at sipforum.org on behalf of amit Sent: Sat 10/6/2007 6:40 PM To: discussion at sipforum.org Subject: [SIPForum-discussion] confrence in sip Hi All, I am working on Sip Chat Sever. Sip can support chat conference?????? Thanks in Advance Amit _______________________________________________ This is the SIP Forum discussion mailing list TO UNSUBSCRIBE, or edit your delivery options, please visit http://sipforum.org/mailman/listinfo/discussion Post to the list at discussion at sipforum.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071007/572faac4/attachment.html From sivam at motorola.com Sun Oct 7 04:26:45 2007 From: sivam at motorola.com (Siva M-Q16748) Date: Sun, 7 Oct 2007 16:26:45 +0800 Subject: [SIPForum-discussion] Out of Office AutoReply: Do we like the same books? Message-ID: <988EE2C769AC284ABAE9328BFC10703F296053@ZMY16EXM66.ds.mot.com> Hi , I am Out Of Office till 09-Oct-2007 . I will be able to reply to your mail only on 10-Oct-2007 . During this time please contact my manager Rao K Venkateswara - Q16395 for any issues. Best Regards, Siva M Mobile : 9880108336 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071007/5968251f/attachment.html From baolovebao at gmail.com Sun Oct 7 07:38:50 2007 From: baolovebao at gmail.com (Donald Lee) Date: Sun, 7 Oct 2007 19:38:50 +0800 Subject: [SIPForum-discussion] Rtp testing tool In-Reply-To: <555d2a660710050047t308cd98dicbf4462c70a68901@mail.gmail.com> References: <555d2a660710050047t308cd98dicbf4462c70a68901@mail.gmail.com> Message-ID: I think ethereal tool can meet part of your requirement. On 10/5/07, darshan b n wrote: > > Hi all, > > I want to test RTP protocol (g.711,g.729..........AMR)..........Can you > please suggest open souce tool which gives statistics such as MOS > ,JITTER,LATENCY....etc > > > -- > Darshan B N > > Thanks & Regards > > _______________________________________________ > This is the SIP Forum discussion mailing list > TO UNSUBSCRIBE, or edit your delivery options, please visit > http://sipforum.org/mailman/listinfo/discussion > Post to the list at discussion at sipforum.org > > -- BR Donald -------------- next part -------------- An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071007/573aa0cb/attachment.html From lingyunxjtu at gmail.com Sun Oct 7 20:40:31 2007 From: lingyunxjtu at gmail.com (Karl Tian) Date: Mon, 8 Oct 2007 08:40:31 +0800 Subject: [SIPForum-discussion] The max duration of SIP conversation In-Reply-To: <200842.58269.qm@web56703.mail.re3.yahoo.com> References: <807efb400710032342rdb8ca9fyc456c7a3ab622060@mail.gmail.com> <200842.58269.qm@web56703.mail.re3.yahoo.com> Message-ID: <807efb400710071740r20fb63aej93c5119380c77d36@mail.gmail.com> Hi badal, Thanks for your reply to my question. Now I've oriented the reason for sip conversation disconnect once duration reaching to 72 hours. The SIP server I used for testing is of ONDO types, and there is a configuration "Talking timeout" with default value "259200000ms", and I will make sure that by capturing packets. Best wishes Your Karl On 10/4/07, badal naik wrote: > > Karl, > as i know there is no special timer that controls a > SIP Session.I have the experience of media session > open over 100 hours. > One thing i can suggest u, Please check your session > by session timing from ethereal capture. > Check how much time each packet is taking for round > trip, how much is jitter value etc etc. > > May be u can get a clue from that. Unless and until I > see the packet capture I can't guess the reason.It is > purely your environment and setting issue.Nothing > related to SIP universal implementation. > > To get the details of timer used in SIP, U can go to > RFC3261 and seach there. > To make ur effort more easy search"Summary of timers". > > Thanks > Badal Naik > --- Karl Tian wrote: > > > Hello everyone, > > Who can tell me if some rfc protocal(for > > example: rfc3261) has define > > the max duration of a sip conversation as 72 hours? > > Now I'm testing the haleness for a kind of sip > > client, but the > > conversations of those clients all stop when the > > duration reachs to 72 > > hours. I guess that the question may be caused by a > > special timer. > > Please help me about this, thanks! > > > > > > > > > > -- > > Karl.Tian > > Infinite Shanghai Communication Terminals Ltd. > > Email :lingyunxjtu at gmail.com > > Msn:lingyunxjtu at hotmail.com > > > _______________________________________________ > > This is the SIP Forum discussion mailing list > > TO UNSUBSCRIBE, or edit your delivery options, > > please visit > > http://sipforum.org/mailman/listinfo/discussion > > Post to the list at discussion at sipforum.org > > > > > > > ____________________________________________________________________________________ > Shape Yahoo! in your own image. Join our Network Research Panel today! > http://surveylink.yahoo.com/gmrs/yahoo_panel_invite.asp?a=7 > > > -- Karl.Tian Infinite Shanghai Communication Terminals Ltd. Mobile:(+86)15902148975 Office :(+86)21-38954999-775 Email :lingyunxjtu at gmail.com Msn:lingyunxjtu at hotmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071008/88e82f90/attachment.html From Richard.Agonias at digitel.ph Sun Oct 7 21:49:41 2007 From: Richard.Agonias at digitel.ph (Agonias, Richard L. (Digitel-GSM)) Date: Mon, 8 Oct 2007 09:49:41 +0800 Subject: [SIPForum-discussion] AMR In-Reply-To: <807efb400710071740r20fb63aej93c5119380c77d36@mail.gmail.com> Message-ID: <90B7FB18EBCD424B83BA9FEA0D4319E8337664@dgtlmail.digitel.ph> Hi All, Good day! I know you guys used AMR already. Does anyone here know if the 39 bits on the AMR SID occurs every 20ms or 160ms? RFC 3267 did not specify it. Thanks! -richard -------------- next part -------------- An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071008/6bb14ed0/attachment.html From govindraj_h at yahoo.co.in Sun Oct 7 23:31:58 2007 From: govindraj_h at yahoo.co.in (Govindraj.B.H @ Gkk) Date: Mon, 8 Oct 2007 09:01:58 +0530 (IST) Subject: [SIPForum-discussion] Asterisk Usage Message-ID: <426133.56377.qm@web8414.mail.in.yahoo.com> Hi Selva, I think there are 2 types of Asterisks One is " * " that means ALL and another is " $ " that means ANY. If any body knows more about this help us. Thanks and Regards Govindraj B H ----- Original Message ---- From: S Selvakumar-VPFR47 To: discussion at sipforum.org Sent: Thursday, 4 October, 2007 2:56:32 AM Subject: [SIPForum-discussion] Asterisk Usage Hi, I want to know some info about Asterisk. Can anyone help me regards Selva Explore your hobbies and interests. Go to http://in.promos.yahoo.com/groups -------------- next part -------------- An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071008/069a49ac/attachment.html From devanand at TechMahindra.com Mon Oct 8 00:51:31 2007 From: devanand at TechMahindra.com (Devanand Kumar) Date: Mon, 8 Oct 2007 10:21:31 +0530 Subject: [SIPForum-discussion] Asterisk Usage Message-ID: <089781E831473740B23334AE52636CD30825A60B@SINBNGEX001.TechMahindra.com> Hi, Attached document will give a brief introduction of Asterisk PBX software. If U want more information u can follow the following link. http://www.digium.com . http://www.asterisk.org/ Thanks and Regards, Devanand Kumar ________________________________ From: discussion-bounces at sipforum.org [mailto:discussion-bounces at sipforum.org] On Behalf Of Govindraj.B.H @ Gkk Sent: Monday, October 08, 2007 9:02 AM To: S Selvakumar-VPFR47; discussion at sipforum.org Subject: Re: [SIPForum-discussion] Asterisk Usage Hi Selva, I think there are 2 types of Asterisks One is " * " that means ALL and another is " $ " that means ANY. If any body knows more about this help us. Thanks and Regards Govindraj B H ----- Original Message ---- From: S Selvakumar-VPFR47 To: discussion at sipforum.org Sent: Thursday, 4 October, 2007 2:56:32 AM Subject: [SIPForum-discussion] Asterisk Usage Hi, I want to know some info about Asterisk. Can anyone help me regards Selva ________________________________ Chat on a cool, new interface. No download required. Click here. ============================================================================================================================ Disclaimer: This message and the information contained herein is proprietary and confidential and subject to the Tech Mahindra policy statement, you may review the policy at http://www.techmahindra.com/Disclaimer.html externally and http://tim.techmahindra.com/Disclaimer.html internally within Tech Mahindra. ============================================================================================================================ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071008/74e186cc/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: asterisk.rtf Type: application/rtf Size: 5035 bytes Desc: asterisk.rtf Url : http://sipforum.org/pipermail/discussion/attachments/20071008/74e186cc/attachment-0001.rtf From wellya at wellya.net Mon Oct 8 01:09:05 2007 From: wellya at wellya.net (Stewart.Zhong) Date: Mon, 8 Oct 2007 13:09:05 +0800 (CST) Subject: [SIPForum-discussion] who can send me about QOS training file? Message-ID: <22616909.1191820145327.JavaMail.postfix@l_010_011_015_034> who can send me about QOS training file? or QOS related specs? Very deelply Thanks!!! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071008/23b335aa/attachment.html From mishra.abhishek at wipro.com Mon Oct 8 02:30:46 2007 From: mishra.abhishek at wipro.com (mishra.abhishek at wipro.com) Date: Mon, 8 Oct 2007 12:00:46 +0530 Subject: [SIPForum-discussion] images with sip messages In-Reply-To: <1191676086.6009.18.camel@amit> References: <1191676086.6009.18.camel@amit> Message-ID: <01949ABCDD38EF4A95B8C347C399D1CE057C5FAB@BLR-EC-MBX02.wipro.com> Through the MESSAGE message you can send any attachment. Just check the RFC2778/2779. ________________________________ From: discussion-bounces at sipforum.org [mailto:discussion-bounces at sipforum.org] On Behalf Of amit Sent: Saturday, October 06, 2007 6:38 PM To: discussion at sipforum.org Subject: [SIPForum-discussion] images with sip messages Hi All, Can we send images with sip messages ? if yes, then how it is possible ? Thanks in Advance Amit -------------- next part -------------- An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071008/d51e98b9/attachment.html From chauhan_delhi at yahoo.com Mon Oct 8 07:03:59 2007 From: chauhan_delhi at yahoo.com (Ramesh Chauhan) Date: Mon, 8 Oct 2007 04:03:59 -0700 (PDT) Subject: [SIPForum-discussion] Asterisk- cisco Call Menager Message-ID: <843093.78935.qm@web34411.mail.mud.yahoo.com> Hi, I am getting "503-Service Not Available" from my CCM(Ciso Call Manager). Ethreal logs of asterisk is attached. We have extensions on my CCM and calls are going and comming perfectly fine to CCM extensions. We are ths error message ony when wants to dial ISD number through CCM. From CCM we are able to make ISD Calls. Can anybody help me . why i am getting this error message. For any other information, please free to revert back. Thanking you in anticipation. Regards Chauhan with regards Ramesh Chauhan --------------------------------- Yahoo! oneSearch: Finally, mobile search that gives answers, not web links. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071008/61e85624/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: sip_call_503.zip Type: application/x-zip-compressed Size: 2058 bytes Desc: 2297908990-sip_call_503.zip Url : http://sipforum.org/pipermail/discussion/attachments/20071008/61e85624/attachment.bin From victor.pascual.avila at gmail.com Mon Oct 8 08:10:32 2007 From: victor.pascual.avila at gmail.com (=?ISO-8859-1?Q?Victor_Pascual_=C1vila?=) Date: Mon, 8 Oct 2007 14:10:32 +0200 Subject: [SIPForum-discussion] who can send me about QOS training file? In-Reply-To: <22616909.1191820145327.JavaMail.postfix@l_010_011_015_034> References: <22616909.1191820145327.JavaMail.postfix@l_010_011_015_034> Message-ID: <618e24240710080510x716997b0yc817ea574072724f@mail.gmail.com> Here you are http://www.ssuet.edu.pk/~amkhan/cisco/Cisco_IP_Telephony_QoS_Design_Guide.pdf Victor Pascual On 08/10/2007, Stewart.Zhong wrote: > > > who can send me about QOS training file? or QOS related specs? > > Very deelply Thanks!!! > _______________________________________________ > This is the SIP Forum discussion mailing list > TO UNSUBSCRIBE, or edit your delivery options, please visit > http://sipforum.org/mailman/listinfo/discussion > Post to the list at discussion at sipforum.org > > From smanickam at velankani.com Mon Oct 8 08:53:39 2007 From: smanickam at velankani.com (Shankar Manickam) Date: Mon, 8 Oct 2007 18:23:39 +0530 Subject: [SIPForum-discussion] CSeq in re-Invite In-Reply-To: <40E89886C8B3B54B98C5291646C591AA01BA21FA@ZMY16EXM67.ds.mot.com> Message-ID: <00d401c809aa$3fb86ac0$0e1c000a@blr.velankani.com> Hi All, I am not seeing that cseq gets incremented for Call hold in the given link http://tech-invite.com/Ti-sip-service-1.html Is there any special condition to get it? With regards, Shankar. -----Original Message----- From: S Selvakumar-VPFR47 [mailto:VPFR47 at motorola.com] Sent: Friday, October 05, 2007 9:03 PM To: Shankar Manickam; discussion at sipforum.org Subject: RE: [SIPForum-discussion] CSeq in re-Invite Hi Shankar, Try a Call Hold scenario, where you can find Cseq gets incremented for Hold Invite and UnHold Invite regards Selva _____ From: discussion-bounces at sipforum.org [mailto:discussion-bounces at sipforum.org] On Behalf Of Shankar Manickam Sent: Friday, October 05, 2007 3:38 PM To: discussion at sipforum.org Subject: [SIPForum-discussion] CSeq in re-Invite Hi, Is there any situation where cseq will get incremented for re-Invite? With Regards, Shankar Manickam -------------- next part -------------- An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071008/98f42097/attachment.html From sumantasen at tataelxsi.co.in Tue Oct 9 05:59:10 2007 From: sumantasen at tataelxsi.co.in (Sumanta Sen) Date: Tue, 9 Oct 2007 15:29:10 +0530 Subject: [SIPForum-discussion] SIGCOMP Message-ID: <003501c80a5b$0a5ce160$f829320a@telxsi.com> Hi All, The TS standards suggest that SDP body should not be compressed. So it means that SIP headers are compressed and message body is not. How to decompress such messages ? How should the receiver distinguish between compressed and uncompressed parts of message? Sumanta The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments contained in it. Contact your Administrator for further information. From anurgupta at yahoo.com Tue Oct 9 09:31:52 2007 From: anurgupta at yahoo.com (anurag gupta) Date: Tue, 9 Oct 2007 06:31:52 -0700 (PDT) Subject: [SIPForum-discussion] CSeq in re-Invite In-Reply-To: <00d401c809aa$3fb86ac0$0e1c000a@blr.velankani.com> Message-ID: <34402.17906.qm@web63706.mail.re1.yahoo.com> Hi The RE-INVITE is generated from the callee side, so it can send any CSeq in its generated INVITE or any other message. But if the Re-INVITe is generated from the caller's end, then CSeq should be incremented. Regards Anurag Shankar Manickam wrote: Hi All, I am not seeing that cseq gets incremented for Call hold in the given link http://tech-invite.com/Ti-sip-service-1.html Is there any special condition to get it? With regards, Shankar. -----Original Message----- From: S Selvakumar-VPFR47 [mailto:VPFR47 at motorola.com] Sent: Friday, October 05, 2007 9:03 PM To: Shankar Manickam; discussion at sipforum.org Subject: RE: [SIPForum-discussion] CSeq in re-Invite Hi Shankar, Try a Call Hold scenario, where you can find Cseq gets incremented for Hold Invite and UnHold Invite regards Selva --------------------------------- From: discussion-bounces at sipforum.org [mailto:discussion-bounces at sipforum.org] On Behalf Of Shankar Manickam Sent: Friday, October 05, 2007 3:38 PM To: discussion at sipforum.org Subject: [SIPForum-discussion] CSeq in re-Invite Hi, Is there any situation where cseq will get incremented for re-Invite? With Regards, Shankar Manickam _______________________________________________ This is the SIP Forum discussion mailing list TO UNSUBSCRIBE, or edit your delivery options, please visit http://sipforum.org/mailman/listinfo/discussion Post to the list at discussion at sipforum.org --------------------------------- Be a better Globetrotter. Get better travel answers from someone who knows. Yahoo! Answers - Check it out. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071009/a69d403b/attachment.html From Adrian.Felea at mtsallstream.com Tue Oct 9 11:24:28 2007 From: Adrian.Felea at mtsallstream.com (Felea, Adrian) Date: Tue, 9 Oct 2007 11:24:28 -0400 Subject: [SIPForum-discussion] Caller identity privacy Message-ID: <2059B5D3BB19464A80466224E5F1B55901C59C9C@TJ1EXB02.mtsallstream.com> I am having issue with When dialing *67 from a PSTN phone (to hide the identity) with a name associated, this gets translated to an "ANONYMOUS" name within the "From" sip header. In this case the identity of the caller is not presented to the called party. When dialing the same from a PSTN phone with no name associated, the "ANONYMOUS" name is not part of the "From" sip header anymore and the identity of the caller is presented to the called party. In the RFC 3261 it states: The From header field allows for a display name. A UAC SHOULD use the display name "Anonymous", along with a syntactically correct, but otherwise meaningless URI (like sip:thisis at anonymous.invalid), if the identity of the client is to remain hidden. If I understand this correctly, does this mean that when calling from a PSTN phone with no name associated (and dialing *67 for privacy), the identity of the caller is still presented to the called party because we do not have an "ANONYMOUS" name within the "From" sip header? Thanks, Adrian -------------- next part -------------- An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071009/35668092/attachment.html From nazeema_guttur at yahoo.com Wed Oct 10 07:06:09 2007 From: nazeema_guttur at yahoo.com (nazeema Tasneem) Date: Wed, 10 Oct 2007 04:06:09 -0700 (PDT) Subject: [SIPForum-discussion] security testing Message-ID: <147034.3651.qm@web37506.mail.mud.yahoo.com> Hi all, can anyone tell me about automated tools for testing security considerations(message flooding, registration hijak, call teardown etc) in SIP. Thanks Nazeema ____________________________________________________________________________________ Building a website is a piece of cake. Yahoo! Small Business gives you all the tools to get online. http://smallbusiness.yahoo.com/webhosting From eric.fiorini at u4eatech.com Wed Oct 10 07:39:09 2007 From: eric.fiorini at u4eatech.com (Eric Fiorini) Date: Wed, 10 Oct 2007 13:39:09 +0200 Subject: [SIPForum-discussion] security testing In-Reply-To: <147034.3651.qm@web37506.mail.mud.yahoo.com> Message-ID: <001401c80b32$2eafe990$ed001cac@u4eatech.com> Try this link ... I don't know if this is what you are looking for ... http://www.cert.org/advisories/CA-2003-06.html Eric Fiorini U4EA Technologies -----Original Message----- From: discussion-bounces at sipforum.org [mailto:discussion-bounces at sipforum.org] On Behalf Of nazeema Tasneem Sent: mercredi 10 octobre 2007 13:06 To: discussion at sipforum.org Subject: [SIPForum-discussion] security testing Hi all, can anyone tell me about automated tools for testing security considerations(message flooding, registration hijak, call teardown etc) in SIP. Thanks Nazeema ____________________________________________________________________________ ________ Building a website is a piece of cake. Yahoo! Small Business gives you all the tools to get online. http://smallbusiness.yahoo.com/webhosting _______________________________________________ This is the SIP Forum discussion mailing list TO UNSUBSCRIBE, or edit your delivery options, please visit http://sipforum.org/mailman/listinfo/discussion Post to the list at discussion at sipforum.org -- U4EA Technologies http://www.u4eatech.com From raymond.jender.ctr at disa.mil Wed Oct 10 10:47:41 2007 From: raymond.jender.ctr at disa.mil (Jender, Raymond C CTR DISA JITC) Date: Wed, 10 Oct 2007 07:47:41 -0700 Subject: [SIPForum-discussion] security testing (UNCLASSIFIED) In-Reply-To: <147034.3651.qm@web37506.mail.mud.yahoo.com> References: <147034.3651.qm@web37506.mail.mud.yahoo.com> Message-ID: <00E9117C300386479B7E2F3CDBF798F601E7EC77@pothia.disanet.disa-u.mil> Classification: UNCLASSIFIED Caveats: NONE Here is a comprehensive list of tools available. Have fun.... http://www.voipsa.org/Resources/tools.php Ray -----Original Message----- From: discussion-bounces at sipforum.org [mailto:discussion-bounces at sipforum.org] On Behalf Of nazeema Tasneem Sent: Wednesday, October 10, 2007 4:06 AM To: discussion at sipforum.org Subject: [SIPForum-discussion] security testing Hi all, can anyone tell me about automated tools for testing security considerations(message flooding, registration hijak, call teardown etc) in SIP. Thanks Nazeema ________________________________________________________________________ ____________ Building a website is a piece of cake. Yahoo! Small Business gives you all the tools to get online. http://smallbusiness.yahoo.com/webhosting _______________________________________________ This is the SIP Forum discussion mailing list TO UNSUBSCRIBE, or edit your delivery options, please visit http://sipforum.org/mailman/listinfo/discussion Post to the list at discussion at sipforum.org Classification: UNCLASSIFIED Caveats: NONE From tmaufer at musecurity.com Wed Oct 10 11:46:55 2007 From: tmaufer at musecurity.com (Thomas Maufer) Date: Wed, 10 Oct 2007 08:46:55 -0700 Subject: [SIPForum-discussion] security testing In-Reply-To: <147034.3651.qm@web37506.mail.mud.yahoo.com> Message-ID: Apologies for the shameless plug. My company makes the Mu-4000 Security Analyzer that (among other things) has about 2,000,000 deeply stateful test cases designed to expose protocol implementation flaws. We can deliver those 2 million test cases over any of 5 transports (UDP, TCP, SSLv2, SSLv3, and TLSv1). Early next year, all those will work over IPv6, making 10 different "transport stacks" for SIP. We also support some IMS options that can affect our SIP test cases. The analyzer is highly automated and can monitor the target any way you like. The analyzer also does response-time profiles for how the invalid test cases we are sending affects the target's ability to respond to valid traffic. We have participated in SIPit events for the last year or so (see you in Beijing?) and after next week will have participated in all three IMS Plugfests (so far!). CT Labs uses the product as well, in their VoIP testing facility. Cheers, ~tom On 10/10/07 4:06 AM, "nazeema Tasneem" wrote: > Hi all, > can anyone tell me about automated tools for testing > security considerations(message flooding, registration > hijak, call teardown etc) in SIP. > > Thanks > Nazeema From raosiponline at gmail.com Thu Oct 11 01:51:16 2007 From: raosiponline at gmail.com (sambasivarao Vemula) Date: Thu, 11 Oct 2007 11:21:16 +0530 Subject: [SIPForum-discussion] security testing In-Reply-To: <147034.3651.qm@web37506.mail.mud.yahoo.com> References: <147034.3651.qm@web37506.mail.mud.yahoo.com> Message-ID: HI, Automation for registration flooding and registration hijak one tool is there i.e SIPPY tool ....is it available for open source or not ,I dont have any idea. Regards Samba On 10/10/07, nazeema Tasneem wrote: > > Hi all, > can anyone tell me about automated tools for testing > security considerations(message flooding, registration > hijak, call teardown etc) in SIP. > > Thanks > Nazeema > > > > > > ____________________________________________________________________________________ > Building a website is a piece of cake. Yahoo! Small Business gives you all > the tools to get online. > http://smallbusiness.yahoo.com/webhosting > _______________________________________________ > This is the SIP Forum discussion mailing list > TO UNSUBSCRIBE, or edit your delivery options, please visit > http://sipforum.org/mailman/listinfo/discussion > Post to the list at discussion at sipforum.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071011/5eac846e/attachment.html From VPFR47 at motorola.com Thu Oct 11 04:03:35 2007 From: VPFR47 at motorola.com (S Selvakumar-VPFR47) Date: Thu, 11 Oct 2007 16:03:35 +0800 Subject: [SIPForum-discussion] Replace header in REFER message Message-ID: <40E89886C8B3B54B98C5291646C591AA01BA29E5@ZMY16EXM67.ds.mot.com> Hi, Does anyone know on what basis the Replace header in REFER is build regards Selva -------------- next part -------------- An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071011/1871de9e/attachment.html From chauhan_delhi at yahoo.com Thu Oct 11 04:40:09 2007 From: chauhan_delhi at yahoo.com (Ramesh Chauhan) Date: Thu, 11 Oct 2007 01:40:09 -0700 (PDT) Subject: [SIPForum-discussion] Extension.conf Message-ID: <332532.78809.qm@web34405.mail.mud.yahoo.com> Hi all, Entery in file is as folllows: SIP.CONF: [7795] type=friend username=7795 secret=7795 host=dynamic port=5060 relaxdtmf=yes dtmfmode=inband canreinvite=no context=outgoing_ISD mailbox=7795 at default disallow=all allow=ulaw EXTENSIONS.CONF: [outgoing_STD] exten => _0Zxxxxxxxxx,2,Dial(Zap/r1/${EXTEN}) exten => _77XX,1,Answer() exten => _77XX,2,Dial(SIP/${EXTEN},30,r) exten => _77XX,3,Hangup() Question: We want, if someome dials any outside number, it will ask for passwd. How to configure extension or any other file in that case ? PLease help me out.... Regards Chauhan --------------------------------- Shape Yahoo! in your own image. Join our Network Research Panel today! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071011/8b687331/attachment.html From Karthik_Ramiya at infosys.com Thu Oct 11 05:26:40 2007 From: Karthik_Ramiya at infosys.com (Karthik Ramiya) Date: Thu, 11 Oct 2007 14:56:40 +0530 Subject: [SIPForum-discussion] Replace header in REFER message In-Reply-To: <40E89886C8B3B54B98C5291646C591AA01BA29E5@ZMY16EXM67.ds.mot.com> References: <40E89886C8B3B54B98C5291646C591AA01BA29E5@ZMY16EXM67.ds.mot.com> Message-ID: <13F04D2767AA2D42878E8ED4856BCB6502B0FA918E@BLRKECMBX02.ad.infosys.com> Hi Selva, Hope RFC3891 helps u. Thanks and regards, Karthik ________________________________ From: discussion-bounces at sipforum.org [mailto:discussion-bounces at sipforum.org] On Behalf Of S Selvakumar-VPFR47 Sent: Thursday, October 11, 2007 1:34 PM To: discussion at sipforum.org Subject: [SIPForum-discussion] Replace header in REFER message Hi, Does anyone know on what basis the Replace header in REFER is build regards Selva **************** CAUTION - Disclaimer ***************** This e-mail contains PRIVILEGED AND CONFIDENTIAL INFORMATION intended solely for the use of the addressee(s). If you are not the intended recipient, please notify the sender by e-mail and delete the original message. Further, you are not to copy, disclose, or distribute this e-mail or its contents to any other person and any such actions are unlawful. This e-mail may contain viruses. Infosys has taken every reasonable precaution to minimize this risk, but is not liable for any damage you may sustain as a result of any virus in this e-mail. You should carry out your own virus checks before opening the e-mail or attachment. Infosys reserves the right to monitor and review the content of all messages sent to or from this e-mail address. Messages sent to or from this e-mail address may be stored on the Infosys e-mail system. ***INFOSYS******** End of Disclaimer ********INFOSYS*** -------------- next part -------------- An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071011/dbc9484c/attachment.html From joel.silva at novabase.pt Thu Oct 11 06:28:24 2007 From: joel.silva at novabase.pt (Joel Silva) Date: Thu, 11 Oct 2007 11:28:24 +0100 Subject: [SIPForum-discussion] Edit sip-communicator layout References: <332532.78809.qm@web34405.mail.mud.yahoo.com> Message-ID: Hello. I would like to edit the layout of the communicator but I find it to complex. Can someone give me some help or advise me some tools to do this? Thanks, Joel. ________________________________ From: discussion-bounces at sipforum.org [mailto:discussion-bounces at sipforum.org] On Behalf Of Ramesh Chauhan Sent: quinta-feira, 11 de Outubro de 2007 9:40 To: discussion at sipforum.org Subject: [SPAM] - [SIPForum-discussion] Extension.conf - Sending mail server found on xbl.spamhaus.org Hi all, Entery in file is as folllows: SIP.CONF: [7795] type=friend username=7795 secret=7795 host=dynamic port=5060 relaxdtmf=yes dtmfmode=inband canreinvite=no context=outgoing_ISD mailbox=7795 at default disallow=all allow=ulaw EXTENSIONS.CONF: [outgoing_STD] exten => _0Zxxxxxxxxx,2,Dial(Zap/r1/${EXTEN}) exten => _77XX,1,Answer() exten => _77XX,2,Dial(SIP/${EXTEN},30,r) exten => _77XX,3,Hangup() Question: We want, if someome dials any outside number, it will ask for passwd. How to configure extension or any other file in that case ? PLease help me out.... Regards Chauhan ________________________________ Shape Yahoo! in your own image. Join our Network Research Panel today! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071011/6646a097/attachment.html From deepanshu at huawei.com Thu Oct 11 21:35:26 2007 From: deepanshu at huawei.com (Deepanshu) Date: Fri, 12 Oct 2007 09:35:26 +0800 Subject: [SIPForum-discussion] Replace header in REFER message References: <40E89886C8B3B54B98C5291646C591AA01BA29E5@ZMY16EXM67.ds.mot.com> Message-ID: <005d01c80c70$2ab032c0$9e78a40a@china.huawei.com> Replace header is based on the dialog information (ID, to-tag and from-tag) of a current dialog. An INVITE with the Replace header will replace the current dialog. For example A is in dialog with B. C send a INVITE with replace (with dialog information) header to A. Dialog between A and B gets replaced by a new dialog between A and C. BR Deepanshu Gautam Huawei Technologies Co. Ltd. ----- Original Message ----- From: S Selvakumar-VPFR47 To: discussion at sipforum.org Sent: Thursday, October 11, 2007 4:03 PM Subject: [SIPForum-discussion] Replace header in REFER message Hi, Does anyone know on what basis the Replace header in REFER is build regards Selva ------------------------------------------------------------------------------ _______________________________________________ This is the SIP Forum discussion mailing list TO UNSUBSCRIBE, or edit your delivery options, please visit http://sipforum.org/mailman/listinfo/discussion Post to the list at discussion at sipforum.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071012/59f4f8eb/attachment.html From jwwlhc at yahoo.com.cn Thu Oct 11 23:19:38 2007 From: jwwlhc at yahoo.com.cn (=?gb2312?q?=CE=AA=20=BD=AA?=) Date: Fri, 12 Oct 2007 11:19:38 +0800 (CST) Subject: [SIPForum-discussion] (no subject) Message-ID: <368464.13422.qm@web15206.mail.cnb.yahoo.com> thanks! --------------------------------- @yahoo.cn ?????????????????????????? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071012/54f0a69b/attachment.html From alexzhang at gdnt.com.cn Fri Oct 12 04:38:12 2007 From: alexzhang at gdnt.com.cn (alexzhang at gdnt.com.cn) Date: Fri, 12 Oct 2007 16:38:12 +0800 Subject: [SIPForum-discussion] Is the non-2xx final response before the PRACK of the previous provisonal response allowed? Message-ID: <8E523FC208B8174790E69947E307914702083C82@rnd-ex01.rnd.gdnt.local> Hi, Here I am wondering if this scenario can be compliant the SIP protocol? UAC UAS --------- Invite --------------> <--------100 Trying--------- <---------183 SP ----------- something happened before the PRACK <---------4xx failure-------- Alex Zhang Guangdong Nortel (GDNT) R&D Center GSM/UMTS Voice Core - MSC Design Phone: (PSTN)+86 020 89188782 (ESN)6 554 8782 E-mail: alexzhang at gdnt.com.cn YahooIM: zcc_nuaa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071012/40698db2/attachment.html From avorlando at yahoo.com Fri Oct 12 08:53:55 2007 From: avorlando at yahoo.com (Anthony Orlando) Date: Fri, 12 Oct 2007 05:53:55 -0700 (PDT) Subject: [SIPForum-discussion] Request timeouts Message-ID: <423261.67192.qm@web51010.mail.re2.yahoo.com> All, I have a question that I can't find an answer for in any RFC etc. My question is as follows. I have a S-CSCF that has done a DNS query for an application server pair that runs in an active standby mode. I am returned an address for two application servers with priority 1 for primary and 100 for secondary. A call is established on AS1 then is failed. AS2 becomes active. When the phone is hung up it sends BYE messages from the S-CSCF to AS1 for five minutes before he tries the secondary. Obviously this is too long. Can anyone reference an RFC that covers this? Thoughts or feelings are also welcome. ____________________________________________________________________________________ Moody friends. Drama queens. Your life? Nope! - their life, your story. Play Sims Stories at Yahoo! Games. http://sims.yahoo.com/ From marc.robins at sipforum.org Fri Oct 12 12:12:31 2007 From: marc.robins at sipforum.org (Marc Robins) Date: Fri, 12 Oct 2007 12:12:31 -0400 Subject: [SIPForum-discussion] The IIT VoIP Conference and EXPO 2007 Message-ID: <005601c80cea$b17f2700$6501a8c0@RCG> Dear SIP Forum Members, The SIP Forum is proud to announce that we are co-sponsoring the upcoming third annual VoIP Conference and EXPO, hosted by the Illinois Institute of Technology in Wheaton, Illinois. Taking place on Thursday and Friday, October 25 and 26, 2007, VoIP Conference and EXPO 2007 features more than 38 sessions from industry experts and a wine and cheese networking event. The full conference agenda is available at http://www.cpd.iit.edu/voipconference07/schedule.php. Registration Information: As part of the Forum?s co-sponsorship, individual Forum members are entitled to attend the conference at the group rate of $125. Conference Registration includes admission to both days of the conference and expo, breakfast and lunch on both days, the wine and cheese reception on Thursday evening, presenter materials and conference tote bag. To register for the event, please visit http://www.cpd.iit.edu/voipconference07/register.php Participating Organizations: Participating organizations include: Alcatel-Lucent, AT&T, Acme Packet, Azaire Networks, BIT, Booz-Allen-Hamilton, Cbeyond, Cimco, Columbia University, Digium-Asterisk, Enabling Technologies, GeckoTech, Informity Networks, IEEE ComSoc, IIT, International Engineering Consortium, Intrado, M5, Morgan Franklin, Motorola, Neustar, Occam, Penn State University, Performance Technologies, Reef Point Systems, SIP Forum, Spirent, Telchemy, Telcordia, Teleprime, YS-41 Communications, Westell and many others. Exhibit/Sponsorship Opportunities: There are still opportunities to exhibit and sponsor the event. Rates include: Gold Sponsor & Exhibitor - $750 (2 free attendees) Silver Sponsor & Exhibitor - $550 (2 free attendees) Exhibitor only - $325 (2 free attendees) Sponsor only - $300 (1 free attendee) For full details, visit http://www.cpd.iit.edu/voipconference07/exhibitorsponsor.php For More Information: Complete information about the conference can be found at the conference website at http://www.cpd.iit.edu/voipconference07. For questions, contact Scott Pfeiffer at pfeiffer at iit.edu or by phone at 630-682-6001. ************************* Marc Robins Managing Director Elect, SIP Forum www.sipforum.org Chief Evangelism Officer, RCG Tel: 718-548-7245 Mobile: 203-829-6307 SkypeMe! marcrobins http://www.robinsconsult.com ************************* -------------- next part -------------- An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071012/003392f0/attachment.html From suba_technical at yahoo.co.in Sat Oct 13 06:42:03 2007 From: suba_technical at yahoo.co.in (Subashini Rajaraman) Date: Sat, 13 Oct 2007 11:42:03 +0100 (BST) Subject: [SIPForum-discussion] Fax-Passthrough in AC48301! Message-ID: <170161.51455.qm@web94415.mail.in2.yahoo.com> Hi All, In fax-pass through What is the rtp payload type in AC48301. Wheather this Dsp supports all the four types of mode. Fax relay Fax passthrough Modem Relay Modem passthrough I guess the Rtp payload type should be the user configured coder or high bit coder(G711) in Fax passthrough mode. Is it Correct? How the Dsp AC48301 detects the CED tone in Fax passthrough mode.what is the modem rate should be used.Can we relay enable any of the modem type in the fax pass through mode. I had a problem in detecting CED tone in AC48301.According to the manual what are the modems should be relayenable and bypass enable in Faxpassthrough mode & why?. If all should be bypassenable then how the Fax messages(CED,training)will be detected by the DSP. --- Thanks in advance, subashini. --------------------------------- Forgot the famous last words? Access your message archive online. Click here. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071013/37bc109e/attachment.html From mohammadfaiz2003 at yahoo.com Sun Oct 14 05:01:51 2007 From: mohammadfaiz2003 at yahoo.com (�The Passenger�) Date: Sun, 14 Oct 2007 02:01:51 -0700 (PDT) Subject: [SIPForum-discussion] (no subject) Message-ID: <537097.29250.qm@web39611.mail.mud.yahoo.com> mohammadfaiz2003 at yahoo.com ((mOhAmAd fAiZ)) --------------------------------- Moody friends. Drama queens. Your life? Nope! - their life, your story. Play Sims Stories at Yahoo! Games. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071014/865b0197/attachment.html From mohammadfaiz2003 at yahoo.com Sun Oct 14 05:19:19 2007 From: mohammadfaiz2003 at yahoo.com (�The Passenger�) Date: Sun, 14 Oct 2007 02:19:19 -0700 (PDT) Subject: [SIPForum-discussion] Questions Message-ID: <185264.68570.qm@web39612.mail.mud.yahoo.com> Hi every one I'm a new member in SIP Forum and I?m master student from Malaysia I?m doing my master in implementation between SIP and another multimedia protocol and I just want to know some issues 1. What is the codec that SIP supports them? 2. Are RTP and RSTP the only protocols that SIP uses them for exchanging media? 3. I know that I need to do signaling translation and media translation as well, so what are the important issues that I need to put it in my consideration? 4. If you have any other information that you think it important and I need to know it please I will appreciate that Best Regards ((mOhAmAd fAiZ)) --------------------------------- Need a vacation? Get great deals to amazing places on Yahoo! Travel. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071014/25b5d935/attachment.html From pinakee.b at xius.com Mon Oct 15 00:12:34 2007 From: pinakee.b at xius.com (Pinakeeb) Date: Mon, 15 Oct 2007 09:42:34 +0530 Subject: [SIPForum-discussion] Request timeouts In-Reply-To: <423261.67192.qm@web51010.mail.re2.yahoo.com> Message-ID: <200710150412.l9F4CC3h022420@serv1.xius.com> Anthony, The scenario mentioned in your mail is from IMS. I don't think there is any RFC for IMS. IMS is defined in 3GPP. You will find the standards there - http://www.3gpp.org Cheers, Pinakee -----Original Message----- From: discussion-bounces at sipforum.org [mailto:discussion-bounces at sipforum.org] On Behalf Of Anthony Orlando Sent: Friday, October 12, 2007 6:24 PM To: discussion at sipforum.org Subject: [SIPForum-discussion] Request timeouts All, I have a question that I can't find an answer for in any RFC etc. My question is as follows. I have a S-CSCF that has done a DNS query for an application server pair that runs in an active standby mode. I am returned an address for two application servers with priority 1 for primary and 100 for secondary. A call is established on AS1 then is failed. AS2 becomes active. When the phone is hung up it sends BYE messages from the S-CSCF to AS1 for five minutes before he tries the secondary. Obviously this is too long. Can anyone reference an RFC that covers this? Thoughts or feelings are also welcome. ____________________________________________________________________________ ________ Moody friends. Drama queens. Your life? Nope! - their life, your story. Play Sims Stories at Yahoo! Games. http://sims.yahoo.com/ _______________________________________________ This is the SIP Forum discussion mailing list TO UNSUBSCRIBE, or edit your delivery options, please visit http://sipforum.org/mailman/listinfo/discussion Post to the list at discussion at sipforum.org From adams-gao at hotmail.com Mon Oct 15 02:54:37 2007 From: adams-gao at hotmail.com (=?gb2312?B?uN+9qA==?=) Date: Mon, 15 Oct 2007 14:54:37 +0800 Subject: [SIPForum-discussion] "Failed to bind to socket 1868" Message-ID: Hi,All: I am a fresh man of using SipXtapi,there are some problems when i build my project with c#. My Server using Asterisk,and X-lite works on it well,but the sipXezphone i complied does not work, it regists failed. i used sipXezphone default settings,it always returns "Failed to bind to socket 1868". and the log writes as fellows,any one can give me a hand? with many thanks! "2007-10-15T06:26:10.949000Z":3:SIPXTAPI:DEBUG:e-5d2790a721fd4::00000000:sipXtapi:"ENTER FUNC (tid=3744) sipxInitialize\n""2007-10-15T06:26:10.949000Z":4:SIPXTAPI:DEBUG:e-5d2790a721fd4::00000000:sipXtapi:"ENTER FUNC (tid=3744) sipxConfigGetVersion\n""2007-10-15T06:26:10.949000Z":5:SIPXTAPI:DEBUG:e-5d2790a721fd4::00000000:sipXtapi:"EXIT FUNC (tid=3744) sipxConfigGetVersion\n""2007-10-15T06:26:10.949000Z":6:SIPXTAPI:INFO:e-5d2790a721fd4::00000000:sipXtapi:"sipXtapi SDK 2.9.1.0 Dbg (built 0000-00-00)""2007-10-15T06:26:10.965000Z":7:SIPXTAPI:INFO:e-5d2790a721fd4::00000000:sipXtapi:"sipxInitialize tcpPort=36004 udpPort=36004 tlsPort=0 rtpPortStart=9000 maxConnections=32 identity=sipx bindTo=0.0.0.0 sequentialPorts=0 certNickname=800, DBLocation=""2007-10-15T06:26:10.965000Z":8:SIPXTAPI:DEBUG:e-5d2790a721fd4::00000000:sipXtapi:"ENTER FUNC (tid=3744) sipxConfigGetAllLocalNetworkIps\n""2007-10-15T06:26:10.965000Z":9:SIPXTAPI:INFO:e-5d2790a721fd4::00000000:sipXtapi:"sipxConfigGetAllLocalNetworkIps index=0 address=168.150.10.241 adapter=eth0""2007-10-15T06:26:10.965000Z":10:SIPXTAPI:DEBUG:e-5d2790a721fd4::00000000:sipXtapi:"EXIT FUNC (tid=3744) sipxConfigGetAllLocalNetworkIps\n""2007-10-15T06:26:10.965000Z":11:SIPXTAPI:INFO:e-5d2790a721fd4::00000000:sipXtapi:"sipXtapi built without NSS support""2007-10-15T06:26:10.965000Z":12:SIP:DEBUG:e-5d2790a721fd4::00000000:sipXtapi:"SipUserAgent::_ sipTcpPort = 36004, sipUdpPort = 36004, sipTlsPort = 0""2007-10-15T06:26:10.965000Z":13:SIP:DEBUG:e-5d2790a721fd4::00000000:sipXtapi:"SipUdpServer::_ port = 36004, bUseNextAvailablePort = 0, szBoundIp = '0.0.0.0'""2007-10-15T06:26:10.980000Z":14:KERNEL:ERR:e-5d2790a721fd4::00000000:sipXtapi:"Failed to bind to socket 1868\n""2007-10-15T06:26:10.980000Z":15:SIP:DEBUG:e-5d2790a721fd4::00000000:sipXtapi:"SipTcpServer::_ port = 36004, taskName = 'SipTcpServer-%d', bUseNextAvailablePort = 0, szBindAddr = '0.0.0.0'""2007-10-15T06:26:10.980000Z":16:KERNEL:DEBUG:e-5d2790a721fd4::00000000:sipXtapi:"OsServerSocket::_ queue=64 port=36004 bindaddr=16 8.150.10.241""2007-10-15T06:26:10.980000Z":17:KERNEL:ERR:e-5d2790a721fd4::00000000:sipXtapi:"OsServerSocket: bind to port 36004 failed with error: 10048 = 0x2740""2007-10-15T06:26:10.996000Z":18:KERNEL:DEBUG:e-5d2790a721fd4::00000000:sipXtapi:"OsTimerTask::getTimerTask OsTimerTask started""2007-10-15T06:26:10.996000Z":19:SIPXTAPI:INFO:e-5d2790a721fd4::00000000:sipXtapi:"Default bind address 168.150.10.241, udpPort=36004, tcpPort=36004, tlsPort=-1""2007-10-15T06:26:10.996000Z":20:SIPXTAPI:INFO:e-5d2790a721fd4::00000000:sipXtapi:"Default Identity: sip:sipx at 168.150.10.241:36004\n""2007-10-15T06:26:11.168000Z":21:KERNEL:DEBUG:e-5d2790a721fd4:NetInTask:000006B0:sipXtapi:"OsServerSocket::_ queue=1 port=-2 bindaddr=127.0.0.1""2007-10-15T06:26:11.168000Z":22:SIP:DEBUG:e-5d2790a721fd4:NetInTaskHelper-15:00000DC0:sipXtapi:"OsConnectionSocket::_ attempt 127.0.0.1:2438 BLOCKING""2007-10-15T06:26:11.168000Z":23:MP:INFO:e-5d2790a721fd4:NetInTaskHelper-15:FFFFFFFF:sipXtapi:"NetInTaskHelper::run()... returning 0, after 1 tries\n""2007-10-15T06:26:11.277000Z":26:CP:DEBUG:e-5d2790a721fd4::00000000:sipXtapi:"ENTER FUNC (tid=3744) CallManager\n""2007-10-15T06:26:11.277000Z":27:CP:DEBUG:e-5d2790a721fd4::00000000:sipXtapi:"EXIT FUNC (tid=3744) CallManager\n""2007-10-15T06:26:11.277000Z":28:SIPXTAPI:DEBUG:e-5d2790a721fd4::00000000:sipXtapi:"ENTER FUNC (tid=3744) sipxConfigSetAudioCodecPreferences\n""2007-10-15T06:26:11.277000Z":29:SIPXTAPI:INFO:e-5d2790a721fd4::00000000:sipXtapi:"sipxConfigSetAudioCodecPreferences hInst=04A45830 bandWidth=2""2007-10-15T06:26:11.277000Z":30:SIPXTAPI:DEBUG:e-5d2790a721fd4::00000000:sipXtapi:"sipxConfigSetAudioCodecPreferences number of Codec = 8 for hInst=04A45830""2007-10-15T06:26:11.277000Z":31:SIPXTAPI:DEBUG:e-5d2790a721fd4::00000000:sipXtapi:"sipxConfigSetAudioCodecPreferences: PCMU PCMA audio/telephone-event SPEEX SPEEX_5 SPEEX_15 SPEEX_24 GSM""2007-10-15T06:26:11.277000Z":32:MP:DEBUG:e-5d2790a721fd4::00000000:sipXtapi:"sipXmediaFactoryImpl::buildCodecFactory: sReferences = PCMU PCMA audio/tele phone-event SPEEX SPEEX_5 SPEEX_15 SPEEX_24 GSM with NumReject 0""2007-10-15T06:26:11.277000Z":33:MP:DEBUG:e-5d2790a721fd4::00000000:sipXtapi:"sipXmediaFactoryImpl::buildCodecFactory: supported codecs = PCMU PCMA audio/telephone-event SPEEX SPEEX_5 SPEEX_15 SPEEX_24 GSM with NumReject 0""2007-10-15T06:26:11.277000Z":34:SIPXTAPI:DEBUG:e-5d2790a721fd4::00000000:sipXtapi:"EXIT FUNC (tid=3744) sipxConfigSetAudioCodecPreferences\n""2007-10-15T06:26:11.277000Z":35:SIPXTAPI:INFO:e-5d2790a721fd4::00000000:sipXtapi:"__sipxEventListenerAdd hInst=04A45830 pCallbackProc=047E1810 pUserData=04ACE618""2007-10-15T06:26:11.777000Z":36:SIPXTAPI:DEBUG:e-5d2790a721fd4::00000000:sipXtapi:"EXIT FUNC (tid=3744) sipxInitialize\n" _________________________________________________________________ Windows Live Spaces ???????????????? http://miaomiaogarden2007.spaces.live.com/ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071015/85775d98/attachment.html From alexzhang at gdnt.com.cn Mon Oct 15 23:08:20 2007 From: alexzhang at gdnt.com.cn (alexzhang at gdnt.com.cn) Date: Tue, 16 Oct 2007 11:08:20 +0800 Subject: [SIPForum-discussion] [Sip-implementors] The non-2xx final response is allowed to besentby UAS before PRACK of the Provisional Response? In-Reply-To: <8E523FC208B8174790E69947E3079147020842D6@rnd-ex01.rnd.gdnt.local> References: <8E523FC208B8174790E69947E307914702083F1B@rnd-ex01.rnd.gdnt.local><0D4E483A0E6E0A46861409E5D6C2011CCF5DFA@sea02-mxc01.citel.com> <8E523FC208B8174790E69947E3079147020842D6@rnd-ex01.rnd.gdnt.local> Message-ID: <8E523FC208B8174790E69947E30791470208432C@rnd-ex01.rnd.gdnt.local> Or does the spec allow the below scenario? UAC UAS |-------Inivte(SDP Offer)--------------->| |<------100 Trying-----------------------| | | |<------183 Session Prog(SDP Anser)------| | | |<------480 Temp Fail--------------------| | | Thanks, Alex 6-554-8782 -----Original Message----- From: sip-implementors-bounces at lists.cs.columbia.edu [mailto:sip-implementors-bounces at lists.cs.columbia.edu] On Behalf Of alexzhang at gdnt.com.cn Sent: Tuesday, October 16, 2007 10:18 AM To: michael.procter at citel.com; sip-implementors at lists.cs.columbia.edu Subject: Re: [Sip-implementors] The non-2xx final response is allowed to besentby UAS before PRACK of the Provisional Response? I am a little about the sentence: " unless the final response is 2xx and any of the unacknowledged reliable provisional responses contained a session description." Is it meaning that if the prvisional response contains the SDP, the UAS must wait for the PRACK before sending the final response? Or, if the provisional response contains the SDP, the UAS must wait for the PRACK before sending the 2xx final response? Thanks, Alex 6-554-8782 -----Original Message----- From: Michael Procter [mailto:michael.procter at citel.com] Sent: Monday, October 15, 2007 4:26 PM To: Alex Zhang (GDNTRND); sip-implementors at lists.cs.columbia.edu Subject: RE: [Sip-implementors] The non-2xx final response is allowed to be sentby UAS before PRACK of the Provisional Response? Yes. RFC 3262, section 3, paragraph 19 (last paragraph in section): The UAS MAY send a final response to the initial request before having received PRACKs for all unacknowledged reliable provisional responses, unless the final response is 2xx and any of the unacknowledged reliable provisional responses contained a session description. In that case, it MUST NOT send a final response until those provisional responses are acknowledged. If the UAS does send a final response when reliable responses are still unacknowledged, it SHOULD NOT continue to retransmit the unacknowledged reliable provisional responses, but it MUST be prepared to process PRACK requests for those outstanding responses. A UAS MUST NOT send new reliable provisional responses (as opposed to retransmissions of unacknowledged ones) after sending a final response to a request. Regards, Michael > -----Original Message----- > From: sip-implementors-bounces at lists.cs.columbia.edu [mailto:sip- > implementors-bounces at lists.cs.columbia.edu] On Behalf Of > alexzhang at gdnt.com.cn > Sent: 15 October 2007 04:15 > To: sip-implementors at lists.cs.columbia.edu > Subject: [Sip-implementors] The non-2xx final response is allowed to > be sentby UAS before PRACK of the Provisional Response? > > Hi, Here I am wondering if this scenario can be compliant to the SIP > protocol? > > UAC UAS > --------- Invite --------------> > <--------100 Trying--------- > <---------183 SP ----------- > something happened > before the PRACK > <---------4xx failure-------- > > > > > Alex Zhang > Guangdong Nortel (GDNT) R&D Center > GSM/UMTS Voice Core - MSC Design > Phone: (PSTN)+86 020 89188782 (ESN)6 554 8782 > E-mail: alexzhang at gdnt.com.cn > YahooIM: zcc_nuaa > > > > > > Alex Zhang > Guangdong Nortel (GDNT) R&D Center > GSM/UMTS Voice Core - MSC Design > Phone: (PSTN)+86 020 89188782 (ESN)6 554 8782 > E-mail: alexzhang at gdnt.com.cn YahooIM: > zcc_nuaa > > > _______________________________________________ > Sip-implementors mailing list > Sip-implementors at lists.cs.columbia.edu > https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors _______________________________________________ Sip-implementors mailing list Sip-implementors at lists.cs.columbia.edu https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors From adams-gao at hotmail.com Wed Oct 17 23:41:24 2007 From: adams-gao at hotmail.com (=?gb2312?B?uN+9qA==?=) Date: Thu, 18 Oct 2007 11:41:24 +0800 Subject: [SIPForum-discussion] sipxezphone register time out Message-ID: Hi,All when i using sipxezphone send register message,the server(asterisk)always returns register failure infomation 401,and the UC seems dosent response it,and it always send register message but no re-register with authorization. refering about the sip pack by Ethereal,i found the UC 's souce port was 2755,and distination port was 5060(default),but the response(trying and unauthorize) distination port was 5060. so i guess the port maybe not correct,and the UC cannt recive response message? can anyone give me a hand and tell me how to solute this problem? thanks!Adams.Gao _________________________________________________________________ MSN ???????????????????????????????????????? http://cn.msn.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071018/809c2eaf/attachment.html From Xiaohong.Xiao at alcatel-sbell.com.cn Thu Oct 18 01:24:23 2007 From: Xiaohong.Xiao at alcatel-sbell.com.cn (XIAO Xiaohong) Date: Thu, 18 Oct 2007 13:24:23 +0800 Subject: [SIPForum-discussion] what's the difference & relations between SIP-I and SIP-T? In-Reply-To: References: Message-ID: -------------- next part -------------- An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071018/7449cc55/attachment.html From zhousiru at gmail.com Thu Oct 18 01:42:12 2007 From: zhousiru at gmail.com (siru zhou) Date: Thu, 18 Oct 2007 13:42:12 +0800 Subject: [SIPForum-discussion] sipxezphone register time out In-Reply-To: References: Message-ID: hi, gao 1. a test using another uac (e.g. Xlite) to see if there's something wrong with your sipxezphone congifguration. 2. is the issue related to NAT ? 3. dump and paste the detail of session packages here BR Siru On 10/18/07, ???? wrote: > > Hi,All > when i using sipxezphone send register message,the > server(asterisk)always returns register failure infomation 401,and the UC > seems dosent response it,and it always send register message but no > re-register with authorization. > refering about the sip pack by Ethereal,i found the UC 's souce port > was 2755,and distination port was 5060(default),but the response(trying and > unauthorize) distination port was 5060. > so i guess the port maybe not correct,and the UC cannt recive response > message? > can anyone give me a hand and tell me how to solute this problem? > > thanks! > Adams.Gao > > ------------------------------ > ???????????????????????????????? Windows Live Mail?? ?????????? > > _______________________________________________ > This is the SIP Forum discussion mailing list > TO UNSUBSCRIBE, or edit your delivery options, please visit > http://sipforum.org/mailman/listinfo/discussion > Post to the list at discussion at sipforum.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071018/f24c91b8/attachment.html From ytian at juniper.net Thu Oct 18 01:47:32 2007 From: ytian at juniper.net (Yong Tian) Date: Thu, 18 Oct 2007 13:47:32 +0800 Subject: [SIPForum-discussion] sipxezphone register time out In-Reply-To: Message-ID: <7B8EBFC47BB8C24F80FF697FC6C7B6DFAA9A54@emailcnrd1.jnpr.net> Hi Adams, What is the VIA header in your request? Could you check? YOng ________________________________ From: discussion-bounces at sipforum.org [mailto:discussion-bounces at sipforum.org] On Behalf Of ???? Sent: 2007??10??18?? 11:41 To: discussion at sipforum.org Subject: [SIPForum-discussion] sipxezphone register time out Hi,All when i using sipxezphone send register message,the server(asterisk)always returns register failure infomation 401,and the UC seems dosent response it,and it always send register message but no re-register with authorization. refering about the sip pack by Ethereal,i found the UC 's souce port was 2755,and distination port was 5060(default),but the response(trying and unauthorize) distination port was 5060. so i guess the port maybe not correct,and the UC cannt recive response message? can anyone give me a hand and tell me how to solute this problem? thanks! Adams.Gao ________________________________ ???????????????????????????????? Windows Live Mail?? ?????????? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071018/46ad5971/attachment.html From jnm_04 at rediffmail.com Thu Oct 18 01:51:56 2007 From: jnm_04 at rediffmail.com (jitendra mohapatra) Date: 18 Oct 2007 05:51:56 -0000 Subject: [SIPForum-discussion] How many dialoge, transaction and call are created Message-ID: <20071018055156.20747.qmail@f5mail12.rediffmail.com> HI everyone I am jitendra newly started SIP kindle anyone tell me how many call ,transaction and dialog created in case of call transfer-attended regards jitendra.. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071018/e29fd53d/attachment.html From kunusan at yahoo.com Thu Oct 18 02:05:01 2007 From: kunusan at yahoo.com (badal naik) Date: Wed, 17 Oct 2007 23:05:01 -0700 (PDT) Subject: [SIPForum-discussion] what's the difference & relations between SIP-I and SIP-T? In-Reply-To: Message-ID: <258184.17290.qm@web56706.mail.re3.yahoo.com> SIP-T stands a general framework for internetworking with ISUP.It sees Sip to carry ISUP as MIME Body. SIP-I:Provides more details about how encapsulation and mapping are to be performed at NNI Interface. Example:when De-encaspulating ISUP message: SIP-T: Sees it as a template overriden by SIP headers. SIP-I:Give details about what information should be taken from ISUP and what to retrive from internetworking fro SIP headers to ISUP parametres. Thanks Badal --- XIAO Xiaohong wrote: > > > _______________________________________________ > This is the SIP Forum discussion mailing list > TO UNSUBSCRIBE, or edit your delivery options, > please visit > http://sipforum.org/mailman/listinfo/discussion > Post to the list at discussion at sipforum.org > ____________________________________________________________________________________ Boardwalk for $500? In 2007? Ha! Play Monopoly Here and Now (it's updated for today's economy) at Yahoo! Games. http://get.games.yahoo.com/proddesc?gamekey=monopolyherenow From sukerry at 126.com Thu Oct 18 02:57:44 2007 From: sukerry at 126.com (Kerry) Date: Thu, 18 Oct 2007 14:57:44 +0800 Subject: [SIPForum-discussion] sipxezphone register time out Message-ID: <471703F0.08A0FC.02526@m15-112.126.com> siru zhou???????? ????please try to set realm fied as "asterisk" ======== 2007-10-18 13:42:12 ???????????????? ======== hi, gao 1. a test using another uac (e.g. Xlite) to see if there's something wrong with your sipxezphone congifguration. 2. is the issue related to NAT ? 3. dump and paste the detail of session packages here BR Siru On 10/18/07, ???? wrote: Hi,All when i using sipxezphone send register message,the server(asterisk)always returns register failure infomation 401,and the UC seems dosent response it,and it always send register message but no re-register with authorization. refering about the sip pack by Ethereal,i found the UC 's souce port was 2755,and distination port was 5060(default),but the response(trying and unauthorize) distination port was 5060. so i guess the port maybe not correct,and the UC cannt recive response message? can anyone give me a hand and tell me how to solute this problem? thanks! Adams.Gao ???????????????????????????????? Windows Live Mail?? ?????????? _______________________________________________ This is the SIP Forum discussion mailing list TO UNSUBSCRIBE, or edit your delivery options, please visit http://sipforum.org/mailman/listinfo/discussion Post to the list at discussion at sipforum.org = = = = = = = = = = = = = = = = = = = = = = ?????????????????? ???? ????????????????????????????Kerry ????????????????????????????sukerry at 126.com ------------------------------------------------- ????????????????????????????,?????? Seawolf Technologies Co. Ltd. Beijing Tel: +86-10-82253150-611 (Office) MSN: suxiangmao at hotmail.com WebSite & Service: http://www.seawolftech.com/ http://www.xrainbow.com.cn/ http://www.17ip.com/ http://www.en400.com/ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071018/01fc0672/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/octet-stream Size: 9062 bytes Desc: not available Url : http://sipforum.org/pipermail/discussion/attachments/20071018/01fc0672/attachment-0001.obj From mrknayak at gmail.com Thu Oct 18 09:29:55 2007 From: mrknayak at gmail.com (Rama krushna Nayak) Date: Thu, 18 Oct 2007 18:59:55 +0530 Subject: [SIPForum-discussion] 5 test case each on PRACK and Conference Message-ID: Hi All, Can anyone tell me 5 test case brifly on each. 1. using PRACK . 2.conference using 3 IP phone. Regards Ramakrushna -------------- next part -------------- An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071018/143702f2/attachment.html From mmostafa at nile-online.net Thu Oct 18 10:57:06 2007 From: mmostafa at nile-online.net (Mostafa Ali) Date: Thu, 18 Oct 2007 16:57:06 +0200 Subject: [SIPForum-discussion] Free CDR Server / Collector References: Message-ID: <026a01c81197$26793270$e2408c3e@engteam565> Dear All , Can anybody recommend a free CDR Serve . BR Mostafa Ali From joel.silva at novabase.pt Thu Oct 18 11:49:56 2007 From: joel.silva at novabase.pt (Joel Silva) Date: Thu, 18 Oct 2007 16:49:56 +0100 Subject: [SIPForum-discussion] Recommend sip/simple client sdk References: <026a01c81197$26793270$e2408c3e@engteam565> Message-ID: Can anyone recommend me a SDK/Simple client sdk? Preferably free, if not a trial version. Thanks, Joel From usman at my.web.pk Thu Oct 18 15:32:21 2007 From: usman at my.web.pk (Usman Rauf) Date: Fri, 19 Oct 2007 00:32:21 +0500 Subject: [SIPForum-discussion] Recommend sip/simple client sdk In-Reply-To: Message-ID: <200710181932.l9IJW0ag026144@sipforum.org> Hello, I need to implement SIP client and server in .Net. I have gone through the basics of SIP architecture but I am not getting a point to take a start in coding .. Can someone guide me on this plz? I'll be extremely thankful Regards, Usman Rauf. -----Original Message----- From: discussion-bounces at sipforum.org [mailto:discussion-bounces at sipforum.org] On Behalf Of Joel Silva Sent: Thursday, October 18, 2007 8:50 PM To: discussion at sipforum.org Subject: [SIPForum-discussion] Recommend sip/simple client sdk Can anyone recommend me a SDK/Simple client sdk? Preferably free, if not a trial version. Thanks, Joel _______________________________________________ This is the SIP Forum discussion mailing list TO UNSUBSCRIBE, or edit your delivery options, please visit http://sipforum.org/mailman/listinfo/discussion Post to the list at discussion at sipforum.org CONFIDENTIALITY NOTICE: This e-mail and its attachments (if any) contain information that is privileged, confidential and subject to legal restrictions and penalties regarding its unauthorized disclosure or other use. The information contained herein is the property of F3 Technologies. It is for the exclusive use of the intended recipient and may not be copied in any way, shape or form, or transmitted in any manner to any other distributor, individual, company or corporation. From mohammadfaiz2003 at yahoo.com Thu Oct 18 22:22:03 2007 From: mohammadfaiz2003 at yahoo.com (�The Passenger�) Date: Thu, 18 Oct 2007 19:22:03 -0700 (PDT) Subject: [SIPForum-discussion] CODEC Message-ID: <460087.92365.qm@web39607.mail.mud.yahoo.com> Hi every body is any one can tell me what is the codec which are already supported by SIP ? Audio and Video Codec also, i know its alot but please if you can memorize some of them thanks alot ((mOhAmAd fAiZ)) __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071018/7710f1aa/attachment.html From yongjie.fan at gmail.com Thu Oct 18 22:49:10 2007 From: yongjie.fan at gmail.com (fan yongjie) Date: Fri, 19 Oct 2007 10:49:10 +0800 Subject: [SIPForum-discussion] Recommend sip/simple client sdk In-Reply-To: References: <026a01c81197$26793270$e2408c3e@engteam565> Message-ID: <73f4c8ed0710181949s1dc526c7l41d9da851d30fe0f@mail.gmail.com> you can find what you need from the link www.SIPfoundry.org 2007/10/18, Joel Silva : > > > Can anyone recommend me a SDK/Simple client sdk? Preferably free, if not > a trial version. > Thanks, > Joel > > _______________________________________________ > This is the SIP Forum discussion mailing list > TO UNSUBSCRIBE, or edit your delivery options, please visit > http://sipforum.org/mailman/listinfo/discussion > Post to the list at discussion at sipforum.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071019/83edb95a/attachment.html From yongjie.fan at gmail.com Thu Oct 18 23:03:19 2007 From: yongjie.fan at gmail.com (fan yongjie) Date: Fri, 19 Oct 2007 11:03:19 +0800 Subject: [SIPForum-discussion] CODEC In-Reply-To: <460087.92365.qm@web39607.mail.mud.yahoo.com> References: <460087.92365.qm@web39607.mail.mud.yahoo.com> Message-ID: <73f4c8ed0710182003tf7401dbxa9c297c62dd00718@mail.gmail.com> SIP does not support codecs directly. If you want to support some codecs, you can try to get some free project to implement it. for example, pjmedia the normal codecs include, G.711u/a, G.729, etc. 2007/10/19, ?The Passenger? : > > Hi every body is any one can tell me what is the codec which are already > supported by SIP ? Audio and Video Codec also, i know its alot but please > if you can memorize some of them > > thanks alot > > > *((mOhAmAd fAiZ))* > > __________________________________________________ > Do You Yahoo!? > Tired of spam? Yahoo! Mail has the best spam protection around > http://mail.yahoo.com > > _______________________________________________ > This is the SIP Forum discussion mailing list > TO UNSUBSCRIBE, or edit your delivery options, please visit > http://sipforum.org/mailman/listinfo/discussion > Post to the list at discussion at sipforum.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071019/5b331d0d/attachment.html From Richard.Agonias at digitel.ph Thu Oct 18 23:35:28 2007 From: Richard.Agonias at digitel.ph (Agonias, Richard L. (Digitel-GSM)) Date: Fri, 19 Oct 2007 11:35:28 +0800 Subject: [SIPForum-discussion] SIP and SDP In-Reply-To: <73f4c8ed0710182003tf7401dbxa9c297c62dd00718@mail.gmail.com> Message-ID: <90B7FB18EBCD424B83BA9FEA0D4319E84095E0@dgtlmail.digitel.ph> Hi, What's the relation between SDP and SIP? Does anyone here have the complete layer stack? Thanks! -chad -------------- next part -------------- An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071019/aa22af94/attachment.html From deveshbissa at rediffmail.com Fri Oct 19 01:44:33 2007 From: deveshbissa at rediffmail.com (devesh bissa) Date: 19 Oct 2007 05:44:33 -0000 Subject: [SIPForum-discussion] Re:CODEC Message-ID: <20071019054433.11267.qmail@f5mail16.rediffmail.com> Hi,SIP is purely signalling protocol,and we use codec for media protocol. Is it correct?Please verify me.Thank you,Devesh Devesh Bissa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071019/12e838ac/attachment.html From jeancosta at gmail.com Fri Oct 19 07:39:09 2007 From: jeancosta at gmail.com (Jean Rodrigo) Date: Fri, 19 Oct 2007 08:39:09 -0300 Subject: [SIPForum-discussion] Instructions about VoiceRD - Asterisk Message-ID: Hi everybody! Does anyone have any experience with the software VoiceRD? I'm trying to install this and create an enviroment to take the authentication process (register) out of asterisk and do it at a LDAP directory. I'll appreciate any kind of help! Thank you! Jean Costa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071019/f1871b47/attachment.html From inako at abcom.al Fri Oct 19 08:07:23 2007 From: inako at abcom.al (Ilir Nako) Date: Fri, 19 Oct 2007 14:07:23 +0200 Subject: [SIPForum-discussion] How to implement callshop References: Message-ID: <006001c81248$9bcd56d0$0a464e50@IlirNako> Hi all I want to implement callshop .I'm serching in internet but have not decided how to implemnt it. I need some help eg: the billing software the voip gateway a server when i connect the accounts of my customers and to use this soft like proxy to use my account with one voip service provider. ??? Best regards Ilir -------------- next part -------------- An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071019/c619a558/attachment.html From Dan.Broussard at Level3.com Fri Oct 19 08:48:54 2007 From: Dan.Broussard at Level3.com (Broussard, Dan) Date: Fri, 19 Oct 2007 06:48:54 -0600 Subject: [SIPForum-discussion] How to implement callshop Message-ID: <5EDAA8E0E2355946B2296BFC33307F851A8F5C@idc1exc0006.corp.global.level3.com> Voip provider http://www.reignmaker.net ----- Original Message ----- From: discussion-bounces at sipforum.org To: discussion at sipforum.org Sent: Fri Oct 19 06:07:23 2007 Subject: [SIPForum-discussion] How to implement callshop Hi all I want to implement callshop .I'm serching in internet but have not decided how to implemnt it. I need some help eg: the billing software the voip gateway a server when i connect the accounts of my customers and to use this soft like proxy to use my account with one voip service provider. ??? Best regards Ilir From victor.pascual.avila at gmail.com Fri Oct 19 11:03:01 2007 From: victor.pascual.avila at gmail.com (=?ISO-8859-1?Q?Victor_Pascual_=C1vila?=) Date: Fri, 19 Oct 2007 17:03:01 +0200 Subject: [SIPForum-discussion] How to implement callshop In-Reply-To: <5EDAA8E0E2355946B2296BFC33307F851A8F5C@idc1exc0006.corp.global.level3.com> References: <5EDAA8E0E2355946B2296BFC33307F851A8F5C@idc1exc0006.corp.global.level3.com> Message-ID: <618e24240710190803k2226cbf5l19ba33055b294d5f@mail.gmail.com> I suggest you to contract a TelephonyServiceProvider (eg. www.voztele.com) and outsource all the VoIP platform if you are not an expert in that. Due most calls will be PSTN routed, it has no sense in deploying your own SIP platform. You could ask you TSP to provide you CPE equipment (for example Linksys PAP2) with reverse polarity enabled. Then, you can connect a billing device between the PAP2 and the analogic phone. This could be the scenario: phone---billing equipment---LinksysPAP2---router---internet---TelephonyOverIP Service Prov I hope it was useful, Kind regards, Victor Pascual On 19/10/2007, Broussard, Dan wrote: > Voip provider http://www.reignmaker.net > > > > > ----- Original Message ----- > From: discussion-bounces at sipforum.org > To: discussion at sipforum.org > Sent: Fri Oct 19 06:07:23 2007 > Subject: [SIPForum-discussion] How to implement callshop > > Hi all > > I want to implement callshop .I'm serching in internet but have not decided how to implemnt it. > I need some help eg: > > the billing software > the voip gateway > a server when i connect the accounts of my customers and to use this soft like proxy to use my account with one > voip service provider. > ??? > > Best regards Ilir > > _______________________________________________ > This is the SIP Forum discussion mailing list > TO UNSUBSCRIBE, or edit your delivery options, please visit http://sipforum.org/mailman/listinfo/discussion > Post to the list at discussion at sipforum.org > From jacksont at binarytelecom.com Fri Oct 19 11:32:00 2007 From: jacksont at binarytelecom.com (Tim Jackson) Date: Fri, 19 Oct 2007 08:32:00 -0700 Subject: [SIPForum-discussion] Instructions about VoiceRD - Asterisk In-Reply-To: Message-ID: <0MKpCa-1IitpO20C0-0004oj@mrelay.perfora.net> www.infoarch.com These guys are Asterisk consultants who will consult over the phone, and have reasonable rates. Tim Jackson Binary Telecom, Inc. 3300 NW 185th #197 Portland, OR 97229 www.binarytelecom.com jacksont at binarytelecom.com 800.594.3670 (toll-free) 503.268.0287 (Direct) 503.564.4534 (fax) _____ From: discussion-bounces at sipforum.org [mailto:discussion-bounces at sipforum.org] On Behalf Of Jean Rodrigo Sent: Friday, October 19, 2007 4:39 AM To: discussion at sipforum.org Subject: [SIPForum-discussion] Instructions about VoiceRD - Asterisk Hi everybody! Does anyone have any experience with the software VoiceRD? I'm trying to install this and create an enviroment to take the authentication process (register) out of asterisk and do it at a LDAP directory. I'll appreciate any kind of help! Thank you! Jean Costa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071019/fb745771/attachment.html From stephen.mcvarnock at aepona.com Fri Oct 19 11:34:38 2007 From: stephen.mcvarnock at aepona.com (Stephen McVarnock) Date: Fri, 19 Oct 2007 16:34:38 +0100 Subject: [SIPForum-discussion] Route header and 'orig' parameter Message-ID: <4718CE8E.2020502@aepona.com> Hi folks, Trying to find details on the use of the 'orig' parameter in the Route header for IMS (AS -> S-CSCF). TS 24.229, section 5.7.3 covers it at a high level but I'm looking for something with more detail on the use of the orig/term parameter. Thanks in advance, Steve. From juan.freitas at novabase.pt Fri Oct 19 12:16:37 2007 From: juan.freitas at novabase.pt (Juan Freitas) Date: Fri, 19 Oct 2007 17:16:37 +0100 Subject: [SIPForum-discussion] SIP/SIMPLE client Message-ID: Hi everyone. I need a SIP/SIMPLE client source code to perform some tests. I just want a very simple client that can see other contacts status. Anyone has one that can share? I know that sip-communicator and do this but I need a simpler client to test a combination of changes to the code rapidly. Thanks, Juan Freitas Analyst ........................................................................ ..................................... Novabase Av. Eng. Duarte Pacheco, 15F . 1099-078 Lisboa - Portugal Tel. (+351) 213 836 300 . Fax (+351) 213 836 301 . mailto:juan.freitas at novabase.pt www.novabase.pt -------------- next part -------------- An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071019/31096946/attachment.html From jonathanpwagner at gmail.com Fri Oct 19 12:42:58 2007 From: jonathanpwagner at gmail.com (Jonathan Wagner) Date: Fri, 19 Oct 2007 12:42:58 -0400 Subject: [SIPForum-discussion] Route header and 'orig' parameter In-Reply-To: <4718CE8E.2020502@aepona.com> References: <4718CE8E.2020502@aepona.com> Message-ID: The orig and term headers indicate to the AS what features should be applied (originating or terminating). So the S-CSCF sends the call to the AS that will apply orginating services and signifies that with 'orig' (i.e. Calling Line ID, call restrictions, etc.), the call is sent back down to the s-CSCF that will continue running the originating service triggers, once complete, the CSCF will begin to run terminating triggers and if necessary send the call to the AS that will apply terminating services and signify that leg as 'term'. On 10/19/07, Stephen McVarnock wrote: > > Hi folks, > > Trying to find details on the use of the 'orig' parameter in the Route > header > for IMS (AS -> S-CSCF). > > TS 24.229, section 5.7.3 covers it at a high level but I'm looking for > something > with more detail on the use of the orig/term parameter. > > Thanks in advance, > Steve. > _______________________________________________ > This is the SIP Forum discussion mailing list > TO UNSUBSCRIBE, or edit your delivery options, please visit > http://sipforum.org/mailman/listinfo/discussion > Post to the list at discussion at sipforum.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071019/7bb30052/attachment.html From govindraj_h at yahoo.co.in Sun Oct 21 21:04:25 2007 From: govindraj_h at yahoo.co.in (Govindraj.B.H @ Gkk) Date: Mon, 22 Oct 2007 06:34:25 +0530 (IST) Subject: [SIPForum-discussion] CODEC Message-ID: <734078.33388.qm@web8414.mail.in.yahoo.com> Yes you are right. SIP is a signaling protocol and codecs are negotiated during the session establishment. These codecs do coding/decoding of RTP packets. All, if I am wrong plz correct me. Regards Govindraj B H ----- Original Message ---- From: devesh bissa To: discussion at sipforum.org Sent: Friday, 19 October, 2007 1:44:33 AM Subject: [SIPForum-discussion] Re:CODEC Hi, SIP is purely signalling protocol,and we use codec for media protocol. Is it correct? Please verify me. Thank you, Devesh Devesh Bissa -----Inline Attachment Follows----- _______________________________________________ This is the SIP Forum discussion mailing list TO UNSUBSCRIBE, or edit your delivery options, please visit http://sipforum.org/mailman/listinfo/discussion Post to the list at discussion at sipforum.org Bring your gang together - do your thing. Go to http://in.promos.yahoo.com/groups -------------- next part -------------- An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071022/7d2769a0/attachment.html From govindraj_h at yahoo.co.in Sun Oct 21 21:10:30 2007 From: govindraj_h at yahoo.co.in (Govindraj.B.H @ Gkk) Date: Mon, 22 Oct 2007 06:40:30 +0530 (IST) Subject: [SIPForum-discussion] CODEC Message-ID: <971052.46899.qm@web8403.mail.in.yahoo.com> G711...........audio codec. G729 ( G729a, G729b, G729ab ) -----audio codecs and T.38........Fax codec. ----- Original Message ---- From: fan yongjie To: ?The Passenger? Cc: discussion at sipforum.org Sent: Thursday, 18 October, 2007 11:03:19 PM Subject: Re: [SIPForum-discussion] CODEC SIP does not support codecs directly. If you want to support some codecs, you can try to get some free project to implement it. for example, pjmedia the normal codecs include, G.711u/a, G.729, etc. 2007/10/19, ?The Passenger? : Hi every body is any one can tell me what is the codec which are already supported by SIP ? Audio and Video Codec also, i know its alot but please if you can memorize some of them thanks alot ((mOh AmAd fAiZ )) __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com _______________________________________________ This is the SIP Forum discussion mailing list TO UNSUBSCRIBE, or edit your delivery options, please visit http://sipforum.org/mailman/listinfo/discussion Post to the list at discussion at sipforum.org -----Inline Attachment Follows----- _______________________________________________ This is the SIP Forum discussion mailing list TO UNSUBSCRIBE, or edit your delivery options, please visit http://sipforum.org/mailman/listinfo/discussion Post to the list at discussion at sipforum.org Bring your gang together - do your thing. Go to http://in.promos.yahoo.com/groups -------------- next part -------------- An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071022/a881faa2/attachment.html From GBeith at empirix.com Sun Oct 21 22:09:03 2007 From: GBeith at empirix.com (Beith, Gordon) Date: Sun, 21 Oct 2007 22:09:03 -0400 Subject: [SIPForum-discussion] CODEC In-Reply-To: <971052.46899.qm@web8403.mail.in.yahoo.com> References: <971052.46899.qm@web8403.mail.in.yahoo.com> Message-ID: There are a bunch more than this....just look at all the IETF codec specs that describe the SDP formats/usages. Many of them are wireless codecs. ________________________________ From: discussion-bounces at sipforum.org [mailto:discussion-bounces at sipforum.org] On Behalf Of Govindraj.B.H @ Gkk Sent: Sunday, October 21, 2007 9:11 PM To: fan yongjie; ?The Passenger? Cc: discussion at sipforum.org Subject: Re: [SIPForum-discussion] CODEC G711...........audio codec. G729 ( G729a, G729b, G729ab ) -----audio codecs and T.38.........Fax codec. ----- Original Message ---- From: fan yongjie To: ?The Passenger? Cc: discussion at sipforum.org Sent: Thursday, 18 October, 2007 11:03:19 PM Subject: Re: [SIPForum-discussion] CODEC SIP does not support codecs directly. If you want to support some codecs, you can try to get some free project to implement it. for example, pjmedia the normal codecs include, G.711u/a, G.729, etc. 2007/10/19, ?The Passenger? : Hi every body is any one can tell me what is the codec which are already supported by SIP ? Audio and Video Codec also, i know its alot but please if you can memorize some of them thanks alot ((mOh AmAd fAiZ )) __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com _______________________________________________ This is the SIP Forum discussion mailing list TO UNSUBSCRIBE, or edit your delivery options, please visit http://sipforum.org/mailman/listinfo/discussion Post to the list at discussion at sipforum.org -----Inline Attachment Follows----- _______________________________________________ This is the SIP Forum discussion mailing list TO UNSUBSCRIBE, or edit your delivery options, please visit http://sipforum.org/mailman/listinfo/discussion Post to the list at discussion at sipforum.org ________________________________ Bring your gang together - do your thing. Start your group. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071021/be0cabe2/attachment-0001.html From vendors at tpsoft.com Sun Oct 21 22:24:01 2007 From: vendors at tpsoft.com (Barry Demchak) Date: Sun, 21 Oct 2007 19:24:01 -0700 Subject: [SIPForum-discussion] Calls vs sessions vs dialogs vs transactions Message-ID: <7.0.1.0.2.20071021191710.034309b0@tpsoft.com> Hi, all -- Sorry for such an elementary question. I'm trying to model some aspects of SIP, and I'm not quite sure how calls, sessions, and dialogs relate. (Dialogs are easy ... they're the context for transactions ... and can support a number of them over time.) I have definitely read RFP3261, but it's not quite helpful here. I have also visited the (excellent) tech-info site. Here's what I'm having trouble with: Is a call the same as a session?? Apparently a session can have numerous dialogs ... when can that happen, and what does it mean? Clearly, setting up a dialog has a formal process. Is there a process for constructing a session or a call ... separate from setting up a dialog?? Thanks. From tuanna at avagroup.vn Mon Oct 22 20:37:37 2007 From: tuanna at avagroup.vn (Nguyen Anh Tuan) Date: Tue, 23 Oct 2007 07:37:37 +0700 Subject: [SIPForum-discussion] (no subject) Message-ID: <7BEC1A7B-F1A4-4D4C-A522-59784F164597@avagroup.vn> I want to use a mobicents sip server (https://mobicents.dev.java.net/ base jboss), however the billing module not found. If who used the mobicent, please help me. Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071023/53d613d4/attachment.html From david.calhoun at cinbell.com Mon Oct 22 22:03:12 2007 From: david.calhoun at cinbell.com (david.calhoun at cinbell.com) Date: Mon, 22 Oct 2007 22:03:12 -0400 Subject: [SIPForum-discussion] UC500 Message-ID: Hi, I'm trying to provide SIP trunking to the UC500, which Broadsoft does not have a PCG for yet (due late this year). Can anyone provide what the configs should look like to have SIP trunks working on the UC500? I can't get the UC500 to even register at this point. Thank you, Dave Calhoun Specialist - Integrated Planner Cincinnati Bell Telephone 209 W7th Street Mail Stop 121-425 Cincinnati, OH 45201 Office: 513.565.2441 Mobile: 513.477.0495 E-mail: david.calhoun at cinbell.com The information transmitted is intended only for the person or entity to which it is addressed and may contain confidential and/or privileged material. Any review, retransmission, dissemination or other use of, or taking of any action in reliance upon, this information by persons or entities other than the intended recipient is prohibited. If you receive this in error, please contact the sender and destroy any copies of this document. From voiptraining at gmail.com Tue Oct 23 00:49:16 2007 From: voiptraining at gmail.com (Kumar DN) Date: Tue, 23 Oct 2007 10:19:16 +0530 Subject: [SIPForum-discussion] Announcing VoIP/SIP training program in Hyderabad, INDIA Message-ID: <957f17eb0710222149t4c2ae042i9e35d2d32e1b5146@mail.gmail.com> Hello SIP Lovers, *Announcing VoIP/SIP Conceptual Course:* I hereby announce the Telecom/VoIP/SIP conceptual course for the job seekers. The ideal candidates for the course should belong to Electronics and Communication Engineering, Computer Science Engineering, and Information Technology Engineering. Please find the attached word document, "VoIP-SIP course content". The classroom training will take place at the following address: Ocean Technologies, Flat No. 501, Above City Financial, Opposite side of S.R. Nagar Police Station, Hyderabad ? 500 038, INDIA Mobile : +91-94400-30971 E-mail : voiptraining at gmail.com With warm regards, Kumar. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071023/a5882a03/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: VoIP-SIP course.doc Type: application/msword Size: 42496 bytes Desc: not available Url : http://sipforum.org/pipermail/discussion/attachments/20071023/a5882a03/attachment-0001.doc From asha.g.pillai at gmail.com Tue Oct 23 07:54:56 2007 From: asha.g.pillai at gmail.com (Asha G) Date: 23 Oct 2007 04:54:56 -0700 Subject: [SIPForum-discussion] Friendship Request on Shelfari Message-ID: <200710231155.l9NBt6xm029826@sipforum.org> An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071023/a90e507e/attachment.html From priyank_mvit at rediffmail.com Tue Oct 23 09:18:13 2007 From: priyank_mvit at rediffmail.com (priyank gupta) Date: 23 Oct 2007 13:18:13 -0000 Subject: [SIPForum-discussion] Regarding Darwin Streaming Server Message-ID: <20071023131813.30450.qmail@f5mail-237-207.rediffmail.com> ? Hi, I have darwin streaming server installed in one of the system. Can any body tell me how to stream the data using Darwin Server...? Which codec Darwin Server will support...? Can we use any file for streaming in Darwin Server...? I read any article on net regarding including hints to play any file in darwin server....? can any body explain me what is this hint and how it helps in streaming a file...? and one more thing what is the difference between Darwin Streaming Server and Quick Time Streaming Server.....? Plz refer any document regarding all this....? thanks Regards Priyank Gupta RnD Department LnT Infotech Limited Bangalore -------------- next part -------------- An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071023/e19885e5/attachment.html From rjsparks at nostrum.com Tue Oct 23 10:14:48 2007 From: rjsparks at nostrum.com (Robert Sparks) Date: Tue, 23 Oct 2007 09:14:48 -0500 Subject: [SIPForum-discussion] SIPit 21 registration closes in less than a week Message-ID: <83A5BDBC-AB50-497A-877F-135F7E8C5AD0@nostrum.com> The registration deadline for SIPit 21 is Oct 29, 6 days from now. If you have not already registered, but plan to attend, please register now. (If you can't register until later in the week, drop me a note - I need to get some information for the event setup from you early). RjS From ashoke.k.ghosh at gmail.com Tue Oct 23 10:27:11 2007 From: ashoke.k.ghosh at gmail.com (Ashoke Kumar Ghosh) Date: Tue, 23 Oct 2007 19:57:11 +0530 Subject: [SIPForum-discussion] SIP stack architecture Message-ID: <004d01c81580$d152f090$da31e0dc@ASHOKE> Hi, Can anybody provide some document on sip stack architecture or refer to some link. Best Regards.. ******************************************* Ashoke Kumar Ghosh Mobile : 919324279664 Email: ashoke.k.ghosh at gmail.com ******************************************* -------------- next part -------------- An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071023/d4e6a756/attachment.html From tuanna at avagroup.vn Tue Oct 23 13:09:54 2007 From: tuanna at avagroup.vn (Nguyen Anh Tuan) Date: Wed, 24 Oct 2007 00:09:54 +0700 Subject: [SIPForum-discussion] SIP stack architecture In-Reply-To: <004d01c81580$d152f090$da31e0dc@ASHOKE> Message-ID: You can search on winkipedia! _____ From: discussion-bounces at sipforum.org [mailto:discussion-bounces at sipforum.org] On Behalf Of Ashoke Kumar Ghosh Sent: Tuesday, October 23, 2007 9:27 PM To: discussion at sipforum.org Subject: [SIPForum-discussion] SIP stack architecture Hi, Can anybody provide some document on sip stack architecture or refer to some link. Best Regards.. ******************************************* Ashoke Kumar Ghosh Mobile : 919324279664 Email: ashoke.k.ghosh at gmail.com ******************************************* -------------- next part -------------- An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071024/ca4b35bf/attachment.html From victor.pascual.avila at gmail.com Tue Oct 23 14:17:03 2007 From: victor.pascual.avila at gmail.com (=?ISO-8859-1?Q?Victor_Pascual_=C1vila?=) Date: Tue, 23 Oct 2007 20:17:03 +0200 Subject: [SIPForum-discussion] Regarding Darwin Streaming Server In-Reply-To: <20071023131813.30450.qmail@f5mail-237-207.rediffmail.com> References: <20071023131813.30450.qmail@f5mail-237-207.rediffmail.com> Message-ID: <618e24240710231117v5f70fe59q8030d63958d03f33@mail.gmail.com> Sorry, could you remember us if Darwin supports only RTSP or supports SIP as well? http://developer.apple.com/opensource/server/streaming/index.html Thanks, Victor On 23 Oct 2007 13:18:13 -0000, priyank gupta wrote: > > > > Hi, > > I have darwin streaming server installed in one of the system. > Can any body tell me how to stream the data using Darwin Server...? > Which codec Darwin Server will support...? > > Can we use any file for streaming in Darwin Server...? > I read any article on net regarding including hints to play any file in > darwin server....? > > can any body explain me what is this hint and how it helps in streaming a > file...? > > and one more thing > what is the difference between Darwin Streaming Server and Quick Time > Streaming Server.....? > > Plz refer any document regarding all this....? > > thanks > Regards > > Priyank Gupta > RnD Department > LnT Infotech Limited > Bangalore > > > > _______________________________________________ > This is the SIP Forum discussion mailing list > TO UNSUBSCRIBE, or edit your delivery options, please visit > http://sipforum.org/mailman/listinfo/discussion > Post to the list at discussion at sipforum.org > > From vendors at tpsoft.com Tue Oct 23 16:07:50 2007 From: vendors at tpsoft.com (Barry Demchak) Date: Tue, 23 Oct 2007 13:07:50 -0700 Subject: [SIPForum-discussion] Calls vs sessions vs dialogs vs transactions Message-ID: <7.0.1.0.2.20071023130738.03456928@tpsoft.com> Thanks, Deepanshu ... the draft-ietf-sipping-dialogusage-04 was very informative. From RFC3261 and from the example in http://www.tech-invite.com/Ti-sip-service-11.html, this is what it looks like to me: (Anyone: feel free to jump in) A dialog (1 usage) is used to set up and maintain a session. A session is an exchange of data, including voice or video. The point of the dialog is to set up the session and then stay out of the way. So, the real feed is the session, and the dialog amounts to out-of-band signalling. A dialog can contain multiple transactions, staged serially. A transaction can affect the dialog and/or the session. There isn't a formal definition of the term "call", though a call-id is part of a dialog identifier. The sense of "call" I get from http://www.tech-invite.com/Ti-sip-service-11.html is that it can contain multiple dialogs (as in the case of a dialog between Alice and Bob, and another between Bob and Carol). As for the relationship between dialogs and usages, a dialog can contain one or more usages, and when the last usage closes, the dialog closes, too. Comments?? Deepanshu?? Anyone else?? Thanks. At 02:29 AM 10/23/2007, Deepanshu wrote: >calls and session are same. > >session can have numerous dialogs in the following case: > >A -----INVITE---------->B >A<-----200OK------------B >-------session established--------- >-------INVITE Dialog established -------- >A---------REFER--------->B >A<---------202 Accepted--------B >--------REFER Dialog established with in current session---------- >A <------------BYE------------B >--------------session/refer dialog/invite dialog ends----------------- > >refer to draft-ietf-sipping-dialogusage-04 for more > >BR >Deepanshu Gautam >Huawei Technologies Co. Ltd. > > > >----- Original Message ----- >From: "Barry Demchak" >To: >Sent: Monday, October 22, 2007 10:24 AM >Subject: [SIPForum-discussion] Calls vs sessions vs dialogs vs transactions > > > > Hi, all -- > > > > Sorry for such an elementary question. I'm trying to model some > > aspects of SIP, and I'm not quite sure how calls, sessions, and > > dialogs relate. (Dialogs are easy ... they're the context for > > transactions ... and can support a number of them over time.) > > > > I have definitely read RFP3261, but it's not quite helpful here. I > > have also visited the (excellent) tech-info site. > > > > Here's what I'm having trouble with: > > > > Is a call the same as a session?? > > > > Apparently a session can have numerous dialogs ... when can that > > happen, and what does it mean? > > > > Clearly, setting up a dialog has a formal process. Is there a process > > for constructing a session or a call ... separate from setting up a >dialog?? > > > > Thanks. > > > > > > > > _______________________________________________ > > This is the SIP Forum discussion mailing list > > TO UNSUBSCRIBE, or edit your delivery options, please visit >http://sipforum.org/mailman/listinfo/discussion > > Post to the list at discussion at sipforum.org From deepanshu at huawei.com Tue Oct 23 21:58:32 2007 From: deepanshu at huawei.com (Deepanshu) Date: Wed, 24 Oct 2007 09:58:32 +0800 Subject: [SIPForum-discussion] Calls vs sessions vs dialogs vs transactions References: <7.0.1.0.2.20071021191710.034309b0@tpsoft.com> <004d01c81557$42aa3c50$9e78a40a@china.huawei.com> <7.0.1.0.2.20071023124254.03556f30@tpsoft.com> Message-ID: <004f01c815e1$61b90740$9e78a40a@china.huawei.com> inine line staring in [DG] ----- Original Message ----- From: "Barry Demchak" To: "Deepanshu" Sent: Wednesday, October 24, 2007 4:06 AM Subject: Re: [SIPForum-discussion] Calls vs sessions vs dialogs vs transactions > Thanks, Deepanshu ... the draft-ietf-sipping-dialogusage-04 was very > informative. > > From RFC3261 and from the example in > http://www.tech-invite.com/Ti-sip-service-11.html, this is what it > looks like to me: > > (Anyone: feel free to jump in) > > A dialog (1 usage) is used to set up and maintain a session. > > A session is an exchange of data, including voice or video. The point > of the dialog is to set up the session and then stay out of the way. > So, the real feed is the session, and the dialog amounts to > out-of-band signalling. > > A dialog can contain multiple transactions, staged serially. A > transaction can affect the dialog and/or the session. > > There isn't a formal definition of the term "call", though a call-id > is part of a dialog identifier. The sense of "call" I get from > http://www.tech-invite.com/Ti-sip-service-11.html is that it can > contain multiple dialogs (as in the case of a dialog between Alice > and Bob, and another between Bob and Carol). [DG] as you said there 'call' is not a formal term in SIP, so anyone can put that in his/her own way. According to me 'call' is neither a session nor a dialog rather it is just a user action. > > As for the relationship between dialogs and usages, a dialog can > contain one or more usages, and when the last usage closes, the > dialog closes, too. [DG] that true. I would like to add one point. This type of behaviour is being criticized because of its complex nature. In draft-ietf-sipping-dialogusage-04 it is said to aviod using dialog with multiple usage. The possible solution is to use Target-Dialog header filed (RFC4538) > > Comments?? Deepanshu?? Anyone else?? > > Thanks. > > At 02:29 AM 10/23/2007, Deepanshu wrote: > >calls and session are same. > > > >session can have numerous dialogs in the following case: > > > >A -----INVITE---------->B > >A<-----200OK------------B > >-------session established--------- > >-------INVITE Dialog established -------- > >A---------REFER--------->B > >A<---------202 Accepted--------B > >--------REFER Dialog established with in current session---------- > >A <------------BYE------------B > >--------------session/refer dialog/invite dialog ends----------------- > > > >refer to draft-ietf-sipping-dialogusage-04 for more > > > >BR > >Deepanshu Gautam > >Huawei Technologies Co. Ltd. > > > > > > > >----- Original Message ----- > >From: "Barry Demchak" > >To: > >Sent: Monday, October 22, 2007 10:24 AM > >Subject: [SIPForum-discussion] Calls vs sessions vs dialogs vs transactions > > > > > > > Hi, all -- > > > > > > Sorry for such an elementary question. I'm trying to model some > > > aspects of SIP, and I'm not quite sure how calls, sessions, and > > > dialogs relate. (Dialogs are easy ... they're the context for > > > transactions ... and can support a number of them over time.) > > > > > > I have definitely read RFP3261, but it's not quite helpful here. I > > > have also visited the (excellent) tech-info site. > > > > > > Here's what I'm having trouble with: > > > > > > Is a call the same as a session?? > > > > > > Apparently a session can have numerous dialogs ... when can that > > > happen, and what does it mean? > > > > > > Clearly, setting up a dialog has a formal process. Is there a process > > > for constructing a session or a call ... separate from setting up a > >dialog?? > > > > > > Thanks. > > > > > > > > > > > > _______________________________________________ > > > This is the SIP Forum discussion mailing list > > > TO UNSUBSCRIBE, or edit your delivery options, please visit > >http://sipforum.org/mailman/listinfo/discussion > > > Post to the list at discussion at sipforum.org > From jeancosta at gmail.com Wed Oct 24 10:01:34 2007 From: jeancosta at gmail.com (Jean Rodrigo) Date: Wed, 24 Oct 2007 11:01:34 -0300 Subject: [SIPForum-discussion] VoiceRD and Edirectory 8.8.1 Instalation Message-ID: Hi everybody! I'm trying to develop an environment where I can use Asterisk integrated to an Ldap directory. I found the VoiceRD software that provides it but I'm not having success in the instalation. Does anyone have any tutorial that could help me to install it? At this moment I'm trying to install the Novell Edirectory 8.8.1 but it gives the message below: [root at localhost setup]# ./nds-install -c server -c admutils -u ./nds-install: line 669: [: too many arguments ./nds-install: line 691: [: too many arguments ./nds-install: line 691: [: too many arguments %%% There are no components available to install. %%% Some packages required for the components may be missing in I'll appreciate any kind of help! Thank you!! Jean Costa. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071024/5f38de4d/attachment.html From vendors at tpsoft.com Wed Oct 24 23:46:41 2007 From: vendors at tpsoft.com (Barry Demchak) Date: Wed, 24 Oct 2007 20:46:41 -0700 Subject: [SIPForum-discussion] Calls vs sessions vs dialogs vs transactions In-Reply-To: <004f01c815e1$61b90740$9e78a40a@china.huawei.com> References: <7.0.1.0.2.20071021191710.034309b0@tpsoft.com> <004d01c81557$42aa3c50$9e78a40a@china.huawei.com> <7.0.1.0.2.20071023124254.03556f30@tpsoft.com> <004f01c815e1$61b90740$9e78a40a@china.huawei.com> Message-ID: <7.0.1.0.2.20071024204537.0347c780@tpsoft.com> Thanks, Deepanshu ... A big help! At 06:58 PM 10/23/2007, Deepanshu wrote: >inine line staring in [DG] >----- Original Message ----- >From: "Barry Demchak" >To: "Deepanshu" >Sent: Wednesday, October 24, 2007 4:06 AM >Subject: Re: [SIPForum-discussion] Calls vs sessions vs dialogs vs >transactions > > > > Thanks, Deepanshu ... the draft-ietf-sipping-dialogusage-04 was very > > informative. > > > > From RFC3261 and from the example in > > http://www.tech-invite.com/Ti-sip-service-11.html, this is what it > > looks like to me: > > > > (Anyone: feel free to jump in) > > > > A dialog (1 usage) is used to set up and maintain a session. > > > > A session is an exchange of data, including voice or video. The point > > of the dialog is to set up the session and then stay out of the way. > > So, the real feed is the session, and the dialog amounts to > > out-of-band signalling. > > > > A dialog can contain multiple transactions, staged serially. A > > transaction can affect the dialog and/or the session. > > > > There isn't a formal definition of the term "call", though a call-id > > is part of a dialog identifier. The sense of "call" I get from > > http://www.tech-invite.com/Ti-sip-service-11.html is that it can > > contain multiple dialogs (as in the case of a dialog between Alice > > and Bob, and another between Bob and Carol). >[DG] as you said there 'call' is not a formal term in SIP, so anyone can put >that in his/her own way. According to me 'call' is neither a session nor a >dialog rather it is just a user action. > > > > As for the relationship between dialogs and usages, a dialog can > > contain one or more usages, and when the last usage closes, the > > dialog closes, too. >[DG] that true. I would like to add one point. This type of behaviour is >being criticized because of its complex nature. In >draft-ietf-sipping-dialogusage-04 it is said to aviod using dialog with >multiple usage. The possible solution is to use Target-Dialog header filed >(RFC4538) > > > > Comments?? Deepanshu?? Anyone else?? > > > > Thanks. > > > > At 02:29 AM 10/23/2007, Deepanshu wrote: > > >calls and session are same. > > > > > >session can have numerous dialogs in the following case: > > > > > >A -----INVITE---------->B > > >A<-----200OK------------B > > >-------session established--------- > > >-------INVITE Dialog established -------- > > >A---------REFER--------->B > > >A<---------202 Accepted--------B > > >--------REFER Dialog established with in current session---------- > > >A <------------BYE------------B > > >--------------session/refer dialog/invite dialog ends----------------- > > > > > >refer to draft-ietf-sipping-dialogusage-04 for more > > > > > >BR > > >Deepanshu Gautam > > >Huawei Technologies Co. Ltd. > > > > > > > > > > > >----- Original Message ----- > > >From: "Barry Demchak" > > >To: > > >Sent: Monday, October 22, 2007 10:24 AM > > >Subject: [SIPForum-discussion] Calls vs sessions vs dialogs vs >transactions > > > > > > > > > > Hi, all -- > > > > > > > > Sorry for such an elementary question. I'm trying to model some > > > > aspects of SIP, and I'm not quite sure how calls, sessions, and > > > > dialogs relate. (Dialogs are easy ... they're the context for > > > > transactions ... and can support a number of them over time.) > > > > > > > > I have definitely read RFP3261, but it's not quite helpful here. I > > > > have also visited the (excellent) tech-info site. > > > > > > > > Here's what I'm having trouble with: > > > > > > > > Is a call the same as a session?? > > > > > > > > Apparently a session can have numerous dialogs ... when can that > > > > happen, and what does it mean? > > > > > > > > Clearly, setting up a dialog has a formal process. Is there a process > > > > for constructing a session or a call ... separate from setting up a > > >dialog?? > > > > > > > > Thanks. > > > > > > > > > > > > > > > > _______________________________________________ > > > > This is the SIP Forum discussion mailing list > > > > TO UNSUBSCRIBE, or edit your delivery options, please visit > > >http://sipforum.org/mailman/listinfo/discussion > > > > Post to the list at discussion at sipforum.org > > From priyank_mvit at rediffmail.com Thu Oct 25 08:00:08 2007 From: priyank_mvit at rediffmail.com (priyank gupta) Date: 25 Oct 2007 12:00:08 -0000 Subject: [SIPForum-discussion] Regarding Darwin Streaming Server Message-ID: <20071025120008.7928.qmail@f5mail-237-205.rediffmail.com> I think it will support only RTSP, it wont support SIP.... Note::::just clarify if i m wrong On Tue, 23 Oct 2007 Victor Pascual ?vila wrote : >Sorry, could you remember us if Darwin supports only RTSP or supports >SIP as well? > >http://developer.apple.com/opensource/server/streaming/index.html > >Thanks, >Victor > >On 23 Oct 2007 13:18:13 -0000, priyank gupta > wrote: > > > > > > > > Hi, > > > > I have darwin streaming server installed in one of the system. > > Can any body tell me how to stream the data using Darwin Server...? > > Which codec Darwin Server will support...? > > > > Can we use any file for streaming in Darwin Server...? > > I read any article on net regarding including hints to play any file in > > darwin server....? > > > > can any body explain me what is this hint and how it helps in streaming a > > file...? > > > > and one more thing > > what is the difference between Darwin Streaming Server and Quick Time > > Streaming Server.....? > > > > Plz refer any document regarding all this....? > > > > thanks > > Regards > > > > Priyank Gupta > > RnD Department > > LnT Infotech Limited > > Bangalore > > > > > > > > _______________________________________________ > > This is the SIP Forum discussion mailing list > > TO UNSUBSCRIBE, or edit your delivery options, please visit > > http://sipforum.org/mailman/listinfo/discussion > > Post to the list at discussion at sipforum.org > > > > Regards Priyank Gupta RnD Department LnT Infotech Limited Bangalore -------------- next part -------------- An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071025/c88d8666/attachment.html From tuanna at avagroup.vn Thu Oct 25 10:35:31 2007 From: tuanna at avagroup.vn (Nguyen Anh Tuan) Date: Thu, 25 Oct 2007 21:35:31 +0700 Subject: [SIPForum-discussion] Conferrence SipPhone Message-ID: <2AD50C0F-5874-4057-AB5E-9A1DCC8E8742@avagroup.vn> Hi everybody! Help me? I'm deverlopping a conferrence call and a hold call module of a sipphone using java language. It based a ims-communication project. If you know about these modules, you send to me please. Thanks verymuch //__ TUANNA __// -------------- next part -------------- An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071025/6ac01a0e/attachment.html From wellya at wellya.net Thu Oct 25 10:37:26 2007 From: wellya at wellya.net (wellya) Date: Thu, 25 Oct 2007 22:37:26 +0800 Subject: [SIPForum-discussion] IMS,FMC reated data Message-ID: <200710252237201605994@wellya.net> VoIP NGN IMS/FMC R&D Testing Scripts DB http://www.wellya.net There are a lot of related E-books for download freely! wellya 2007-10-25 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071025/e12383ef/attachment.html From yuval_e2 at yahoo.com Thu Oct 25 14:32:03 2007 From: yuval_e2 at yahoo.com (yy yy) Date: Thu, 25 Oct 2007 11:32:03 -0700 (PDT) Subject: [SIPForum-discussion] detect duplicate registration Message-ID: <678838.56143.qm@web30615.mail.mud.yahoo.com> Hi, I'm developing a SIP regsitrar server, and I endcountered the following problem: 2 User agents have the same configuration - same user ID, same authentication user, same password configured. Both of them try to register to the same registrar at about the same time. Since their configuration is correct by the registrar database, both of them can be accepted. How can detect such a situation, and accept only one? It is possible that there is only one UA, which reboots and sends me REGISTER messages again & again, possibly with a different ip address in contact (in case of DHCP) on each REGISTER message, in this case I should accept each REGISTER message, since this is a normal situation, which means that if one UA is already registered, and the expiration time has not passed, I should not reject another REGISTER message from the same UA. Is there a way described in the SIP RFCs that overcomes this situation? Regards, Yuval __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com From gzweig at sonusnet.com Fri Oct 26 09:21:13 2007 From: gzweig at sonusnet.com (Zweig, Greg) Date: Fri, 26 Oct 2007 09:21:13 -0400 Subject: [SIPForum-discussion] Parent Child registration vs Wildcard registration In-Reply-To: <957f17eb0710222149t4c2ae042i9e35d2d32e1b5146@mail.gmail.com> References: <957f17eb0710222149t4c2ae042i9e35d2d32e1b5146@mail.gmail.com> Message-ID: <033458F56EC2A64E8D2D7B759FA3E7E73F8598@sonusmail04.sonusnet.com> All, I was curious to get forum members' opinion on the two prevalent methods of registering users that are behind a PBX. Some carriers use a parent child registration that creates a concept of an "alpha" subscriber while others create a wild card so that subscribers within a group can more easily be registered together. I believe the latter was created by Cisco Using the parent child method, the PBX creates a binding between one of its phone numbers as the address of record (AoR) and Contact-URI in the REGISTER message. The registrar understands that a single AoR actually represents many addresses, and so it registers them implicitly. Using the wild card a "." in the address serves as the wild card - up to 32 digits-- so that a variety of users at the same address can be implicitly registered An example might look like Contact: I would appreciate any practical experience forum members have that would provide advantages or disadvantages for the two methods. Thanks, Greg -------------- next part -------------- An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071026/aee56684/attachment.html From mmostafa at nile-online.net Sun Oct 28 17:57:15 2007 From: mmostafa at nile-online.net (Mostafa Ali) Date: Sun, 28 Oct 2007 23:57:15 +0200 Subject: [SIPForum-discussion] Quintum and Echo Cancelation References: <957f17eb0710222149t4c2ae042i9e35d2d32e1b5146@mail.gmail.com> <033458F56EC2A64E8D2D7B759FA3E7E73F8598@sonusmail04.sonusnet.com> Message-ID: <071c01c819ad$81cfe060$d3b435d9@engteam565> Dear All , I have analog Tenor ( 2 FXO ) connected to GSM gateway , when i intiate internet call the called party ( Cellular Phone ) has un-accepted echo . I tried to play with line impedence in the CAS line signalliing configuration with no effect . Can anybody help me solving this serious problem . Thanks & Best Reagrds Mostafa Ali -------------- next part -------------- An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071028/7066963c/attachment.html From test at iphonet.net Mon Oct 29 08:47:57 2007 From: test at iphonet.net (test) Date: Mon, 29 Oct 2007 13:47:57 +0100 Subject: [SIPForum-discussion] PSTN calls becomes silent In-Reply-To: Message-ID: Hi everybody, I noticed that many times (2/4 calls) where an intern PBX is, the line becomes silent during the call, after 6-10 seconds. The called person has a busy signal and the caller doesn?t hear anything. Has anybody idea why? Regards, Zoltan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071029/542023ef/attachment.html From victor.pascual.avila at gmail.com Mon Oct 29 10:14:18 2007 From: victor.pascual.avila at gmail.com (=?ISO-8859-1?Q?Victor_Pascual_=C1vila?=) Date: Mon, 29 Oct 2007 15:14:18 +0100 Subject: [SIPForum-discussion] PSTN calls becomes silent In-Reply-To: References: Message-ID: <618e24240710290714u56d8825fv9897b7393a328d56@mail.gmail.com> Hello, On 29/10/2007, test wrote: > I noticed that many times (2/4 calls) where an intern PBX is, the line > becomes silent during the call, after 6-10 seconds. The called person has a > busy signal and the caller doesn't hear anything. > Has anybody idea why? Is the extension behind a NAT? are you using any nat-traversal system? If possible, attach a ngrep trace at pbx side. Regards, Victor Pascual From aytechmobiles at gmail.com Mon Oct 29 11:10:37 2007 From: aytechmobiles at gmail.com (AYTECH MOBILES) Date: Mon, 29 Oct 2007 15:10:37 +0000 Subject: [SIPForum-discussion] Clock reference abnormal Message-ID: Hi I am a BSS field TECHNNICAIN for HUAWEI equipment I have an alarm like "clock reference abnormal" Could someone help me about this problem? Regards BOLLE -------------- next part -------------- An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071029/74a807d1/attachment.html From ashishdubey1981 at gmail.com Mon Oct 29 11:08:34 2007 From: ashishdubey1981 at gmail.com (ashish dubey) Date: Mon, 29 Oct 2007 08:08:34 -0700 Subject: [SIPForum-discussion] PSTN calls becomes silent In-Reply-To: <618e24240710290714u56d8825fv9897b7393a328d56@mail.gmail.com> References: <618e24240710290714u56d8825fv9897b7393a328d56@mail.gmail.com> Message-ID: <2271ea2f0710290808g3afb93dg24b15d3659903dfc@mail.gmail.com> hi!!, It may be case of NAT, but, first thing that need to check, is echo cancellation is there on server or not. and band width is also responsible for that silence bcoz, i have faced such issue many times. Regards Ashu On 10/29/07, Victor Pascual ?vila wrote: > > Hello, > > On 29/10/2007, test wrote: > > I noticed that many times (2/4 calls) where an intern PBX is, the line > > becomes silent during the call, after 6-10 seconds. The called person > has a > > busy signal and the caller doesn't hear anything. > > Has anybody idea why? > > Is the extension behind a NAT? are you using any nat-traversal system? > > If possible, attach a ngrep trace at pbx side. > > Regards, > Victor Pascual > _______________________________________________ > This is the SIP Forum discussion mailing list > TO UNSUBSCRIBE, or edit your delivery options, please visit > http://sipforum.org/mailman/listinfo/discussion > Post to the list at discussion at sipforum.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071029/7a36b813/attachment.html From mahalfy at hotmail.com Mon Oct 29 11:47:00 2007 From: mahalfy at hotmail.com (Mahmoud El-Alfy) Date: Mon, 29 Oct 2007 17:47:00 +0200 Subject: [SIPForum-discussion] AsteriskNow Message-ID: I have DLINK 4-port FXO, asterisknow sip server, only 3 concurrent calls can run to the fxo, can any body help me to make 4 concurrent calls through the FXO -------------- next part -------------- An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071029/d886908f/attachment.html From tuanna at avagroup.vn Mon Oct 29 23:01:15 2007 From: tuanna at avagroup.vn (Nguyen Anh Tuan) Date: Tue, 30 Oct 2007 10:01:15 +0700 Subject: [SIPForum-discussion] Echo & Noise Cancellation Message-ID: <04CF0E78-C243-4626-98B8-48FF914E9507@avagroup.vn> Hi everybody! Help me, please ! I'm developping a sipphone based IMS-Communicator Project. However, its voice is no good. I want creating Echo and Noise cancellator. If who know, send me email please. Thanks very much! //_________ TUANNA __________// -------------- next part -------------- An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071030/19960afa/attachment.html From achandrashekar at velankani.com Tue Oct 30 02:38:42 2007 From: achandrashekar at velankani.com (Avinash Chandrashekar) Date: Tue, 30 Oct 2007 12:08:42 +0530 Subject: [SIPForum-discussion] Looking for a SIP test tool/equipment to test the SCTP functionality Message-ID: <009501c81abf$835cec80$8a16c580$@com> Hi All, Has anyone know about any test equipment which has the capability to transfer SIP messages on sctp and the platform needed to run it or configure it. Appreciate for all your support. Thanks, Avinash -------------- next part -------------- An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071030/1caeba95/attachment.html From Chris.Gatch at cbeyond.net Tue Oct 30 06:16:50 2007 From: Chris.Gatch at cbeyond.net (Chris Gatch) Date: Tue, 30 Oct 2007 06:16:50 -0400 Subject: [SIPForum-discussion] Parent Child registration vs Wildcardregistration In-Reply-To: <001401c818d8$e9d349c0$6501a8c0@RCG> References: <001401c818d8$e9d349c0$6501a8c0@RCG> Message-ID: <68D2858458D1684999D254CB49C99CFB102A0077@exch-corp01.corp.cbeyond.net> Greg, I have not seen the wildcard example implemented, but I do know that the first approach of using the parent/child is broadly implemented. The source of the concept is the original SIPconnect Interface Specification that was produced before it went through the SIP Forum Technical Working Group. The explicit parent/child language was dropped in favor of a more 'pure' approach. However, we left language that implicitly allowed parent/child while not encouraging it. SIP Application Servers MUST be prepared to accept (but MUST NOT require) registrations for any valid URI that the Service Provider has assigned to an Enterprise. This interface specification does not define any specific action that is triggered by a successful registration; however one possible use of this information might be to update a DNS entry associated with the PBX in a DNS zone managed by the Service Provider. In the case of Cbeyond, for example, we use the registration of the parent user to update the registration information of all SIP users associated with the account that was registered. If you need more information on the way we handle this, feel free to contact me directly, and I can provide some more detail. Chris ________________________________ From: discussion-bounces at sipforum.org [mailto:discussion-bounces at sipforum.org] On Behalf Of Zweig, Greg Sent: Friday, October 26, 2007 9:21 AM To: discussion at sipforum.org Subject: [SIPForum-discussion] Parent Child registration vs Wildcardregistration All, I was curious to get forum members' opinion on the two prevalent methods of registering users that are behind a PBX. Some carriers use a parent child registration that creates a concept of an "alpha" subscriber while others create a wild card so that subscribers within a group can more easily be registered together. I believe the latter was created by Cisco Using the parent child method, the PBX creates a binding between one of its phone numbers as the address of record (AoR) and Contact-URI in the REGISTER message. The registrar understands that a single AoR actually represents many addresses, and so it registers them implicitly. Using the wild card a "." in the address serves as the wild card - up to 32 digits-- so that a variety of users at the same address can be implicitly registered An example might look like Contact: I would appreciate any practical experience forum members have that would provide advantages or disadvantages for the two methods. Thanks, Greg ********************************************************************** This email may contain confidential information. If you are not the intended recipient, please advise by return email and delete immediately without reading or forwarding to others. -- Cbeyond ********************************************************************** -------------- next part -------------- An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071030/beba135a/attachment.html From sreeji.gopal at gmail.com Tue Oct 30 06:32:04 2007 From: sreeji.gopal at gmail.com (Sreeji Gopal) Date: Tue, 30 Oct 2007 16:02:04 +0530 Subject: [SIPForum-discussion] Looking for a SIP test tool/equipment to test the SCTP functionality In-Reply-To: <009501c81abf$835cec80$8a16c580$@com> References: <009501c81abf$835cec80$8a16c580$@com> Message-ID: <77529fb40710300332q111b322do8967cd7ec56ad6dd@mail.gmail.com> Hi Avinash, Are you looking @ some tool that will enable you to test bulk sip calls? -Sreeji On 10/30/07, Avinash Chandrashekar wrote: > > Hi All, > > Has anyone know about any test equipment which has the capability to > transfer SIP messages on sctp and the platform needed to run it or configure > it. > > > > Appreciate for all your support. > > > > Thanks, > Avinash > > _______________________________________________ > This is the SIP Forum discussion mailing list > TO UNSUBSCRIBE, or edit your delivery options, please visit > http://sipforum.org/mailman/listinfo/discussion > Post to the list at discussion at sipforum.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071030/5f1998e3/attachment.html From aytechmobiles at gmail.com Tue Oct 30 07:02:41 2007 From: aytechmobiles at gmail.com (AYTECH MOBILES) Date: Tue, 30 Oct 2007 11:02:41 +0000 Subject: [SIPForum-discussion] LOOKING FOR HUAWEI BSS ENGINEER HELP Message-ID: Hi I am need someone who have already work on HUAWEI BTS 312 (indoor) BTS 3012A(outdoo) BTS 3012(new indoor) AND BTS 3012AE (new outdoor) I need information about the power sytem and power level of these type of BTS Regrads BOLLE -------------- next part -------------- An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071030/ebe67cf6/attachment-0001.html From braveheart_zk at yahoo.com Tue Oct 30 07:44:15 2007 From: braveheart_zk at yahoo.com (Kai Zhang) Date: Tue, 30 Oct 2007 04:44:15 -0700 (PDT) Subject: [SIPForum-discussion] discussion Digest, Vol 27, Issue 37 Message-ID: <336478.33405.qm@web59015.mail.re1.yahoo.com> Hi, can you remove my mail from the mail list? i don't want to receive the mail, thanks a lot! /Kevin ----- Original Message ---- From: "discussion-request at sipforum.org" To: discussion at sipforum.org Sent: Tuesday, October 30, 2007 7:02:42 PM Subject: discussion Digest, Vol 27, Issue 37 Send discussion mailing list submissions to discussion at sipforum.org To subscribe or unsubscribe via the World Wide Web, visit http://sipforum.org/mailman/listinfo/discussion or, via email, send a message with subject or body 'help' to discussion-request at sipforum.org You can reach the person managing the list at discussion-owner at sipforum.org When replying, please edit your Subject line so it is more specific than "Re: Contents of discussion digest..." Today's Topics: 1. Echo & Noise Cancellation (Nguyen Anh Tuan) 2. Looking for a SIP test tool/equipment to test the SCTP functionality (Avinash Chandrashekar) 3. Re: Parent Child registration vs Wildcardregistration (Chris Gatch) 4. Re: Looking for a SIP test tool/equipment to test the SCTP functionality (Sreeji Gopal) 5. LOOKING FOR HUAWEI BSS ENGINEER HELP (AYTECH MOBILES) ---------------------------------------------------------------------- Message: 1 Date: Tue, 30 Oct 2007 10:01:15 +0700 From: "Nguyen Anh Tuan" Subject: [SIPForum-discussion] Echo & Noise Cancellation To: Message-ID: <04CF0E78-C243-4626-98B8-48FF914E9507 at avagroup.vn> Content-Type: text/plain; charset="us-ascii" Hi everybody! Help me, please ! I'm developping a sipphone based IMS-Communicator Project. However, its voice is no good. I want creating Echo and Noise cancellator. If who know, send me email please. Thanks very much! //_________ TUANNA __________// -------------- next part -------------- An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071030/19960afa/attachment-0001.html ------------------------------ Message: 2 Date: Tue, 30 Oct 2007 12:08:42 +0530 From: "Avinash Chandrashekar" Subject: [SIPForum-discussion] Looking for a SIP test tool/equipment to test the SCTP functionality To: Message-ID: <009501c81abf$835cec80$8a16c580$@com> Content-Type: text/plain; charset="us-ascii" Hi All, Has anyone know about any test equipment which has the capability to transfer SIP messages on sctp and the platform needed to run it or configure it. Appreciate for all your support. Thanks, Avinash -------------- next part -------------- An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071030/1caeba95/attachment-0001.html ------------------------------ Message: 3 Date: Tue, 30 Oct 2007 06:16:50 -0400 From: "Chris Gatch" Subject: Re: [SIPForum-discussion] Parent Child registration vs Wildcardregistration To: , "SIP Forum Tech WG" Message-ID: <68D2858458D1684999D254CB49C99CFB102A0077 at exch-corp01.corp.cbeyond.net> Content-Type: text/plain; charset="us-ascii" Greg, I have not seen the wildcard example implemented, but I do know that the first approach of using the parent/child is broadly implemented. The source of the concept is the original SIPconnect Interface Specification that was produced before it went through the SIP Forum Technical Working Group. The explicit parent/child language was dropped in favor of a more 'pure' approach. However, we left language that implicitly allowed parent/child while not encouraging it. SIP Application Servers MUST be prepared to accept (but MUST NOT require) registrations for any valid URI that the Service Provider has assigned to an Enterprise. This interface specification does not define any specific action that is triggered by a successful registration; however one possible use of this information might be to update a DNS entry associated with the PBX in a DNS zone managed by the Service Provider. In the case of Cbeyond, for example, we use the registration of the parent user to update the registration information of all SIP users associated with the account that was registered. If you need more information on the way we handle this, feel free to contact me directly, and I can provide some more detail. Chris ________________________________ From: discussion-bounces at sipforum.org [mailto:discussion-bounces at sipforum.org] On Behalf Of Zweig, Greg Sent: Friday, October 26, 2007 9:21 AM To: discussion at sipforum.org Subject: [SIPForum-discussion] Parent Child registration vs Wildcardregistration All, I was curious to get forum members' opinion on the two prevalent methods of registering users that are behind a PBX. Some carriers use a parent child registration that creates a concept of an "alpha" subscriber while others create a wild card so that subscribers within a group can more easily be registered together. I believe the latter was created by Cisco Using the parent child method, the PBX creates a binding between one of its phone numbers as the address of record (AoR) and Contact-URI in the REGISTER message. The registrar understands that a single AoR actually represents many addresses, and so it registers them implicitly. Using the wild card a "." in the address serves as the wild card - up to 32 digits-- so that a variety of users at the same address can be implicitly registered An example might look like Contact: I would appreciate any practical experience forum members have that would provide advantages or disadvantages for the two methods. Thanks, Greg ********************************************************************** This email may contain confidential information. If you are not the intended recipient, please advise by return email and delete immediately without reading or forwarding to others. -- Cbeyond ********************************************************************** -------------- next part -------------- An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071030/beba135a/attachment-0001.html ------------------------------ Message: 4 Date: Tue, 30 Oct 2007 16:02:04 +0530 From: "Sreeji Gopal" Subject: Re: [SIPForum-discussion] Looking for a SIP test tool/equipment to test the SCTP functionality To: "Avinash Chandrashekar" Cc: discussion at sipforum.org Message-ID: <77529fb40710300332q111b322do8967cd7ec56ad6dd at mail.gmail.com> Content-Type: text/plain; charset="iso-8859-1" Hi Avinash, Are you looking @ some tool that will enable you to test bulk sip calls? -Sreeji On 10/30/07, Avinash Chandrashekar wrote: > > Hi All, > > Has anyone know about any test equipment which has the capability to > transfer SIP messages on sctp and the platform needed to run it or configure > it. > > > > Appreciate for all your support. > > > > Thanks, > Avinash > > _______________________________________________ > This is the SIP Forum discussion mailing list > TO UNSUBSCRIBE, or edit your delivery options, please visit > http://sipforum.org/mailman/listinfo/discussion > Post to the list at discussion at sipforum.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071030/5f1998e3/attachment-0001.html ------------------------------ Message: 5 Date: Tue, 30 Oct 2007 11:02:41 +0000 From: "AYTECH MOBILES" Subject: [SIPForum-discussion] LOOKING FOR HUAWEI BSS ENGINEER HELP To: discussion at sipforum.org Message-ID: Content-Type: text/plain; charset="iso-8859-1" Hi I am need someone who have already work on HUAWEI BTS 312 (indoor) BTS 3012A(outdoo) BTS 3012(new indoor) AND BTS 3012AE (new outdoor) I need information about the power sytem and power level of these type of BTS Regrads BOLLE -------------- next part -------------- An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071030/ebe67cf6/attachment.html ------------------------------ _______________________________________________ This is the SIP Forum discussion mailing list TO UNSUBSCRIBE, or edit your delivery options, please visit http://sipforum.org/mailman/listinfo/discussion Post to the list at discussion at sipforum.org End of discussion Digest, Vol 27, Issue 37 ****************************************** __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071030/dae3ca4c/attachment.html From mmostafa at nile-online.net Tue Oct 30 08:10:03 2007 From: mmostafa at nile-online.net (Mostafa Ali) Date: Tue, 30 Oct 2007 14:10:03 +0200 Subject: [SIPForum-discussion] Quintum and Echo Cancelation References: <957f17eb0710222149t4c2ae042i9e35d2d32e1b5146@mail.gmail.com><033458F56EC2A64E8D2D7B759FA3E7E73F8598@sonusmail04.sonusnet.com> <071c01c819ad$81cfe060$d3b435d9@engteam565> Message-ID: <011501c81aed$cd42d660$e2408c3e@engteam565> Dear All , When I use analog phone directly , the echo isn't noticeable however when I connect it to the Quintum FXO port the echo becomes bad . Can anyone advise abt the needed configuration on the quintum . Thanks BR Mostafa Ali ----- Original Message ----- From: Mostafa Ali To: discussion at sipforum.org Sent: Sunday, October 28, 2007 11:57 PM Subject: [SIPForum-discussion] Quintum and Echo Cancelation Dear All , I have analog Tenor ( 2 FXO ) connected to GSM gateway , when i intiate internet call the called party ( Cellular Phone ) has un-accepted echo . I tried to play with line impedence in the CAS line signalliing configuration with no effect . Can anybody help me solving this serious problem . Thanks & Best Reagrds Mostafa Ali _______________________________________________ This is the SIP Forum discussion mailing list TO UNSUBSCRIBE, or edit your delivery options, please visit http://sipforum.org/mailman/listinfo/discussion Post to the list at discussion at sipforum.org From esampaolesi at alcatel-lucent.com Tue Oct 30 12:26:48 2007 From: esampaolesi at alcatel-lucent.com (SAMPAOLESI, Emiliano (Emiliano)) Date: Tue, 30 Oct 2007 17:26:48 +0100 Subject: [SIPForum-discussion] (no subject) Message-ID: <1D2BEFA1C2F6BE4AA36B5FAEF07083E97AB314@DEEXC1U03.de.lucent.com> Emiliano Sampaolesi Alcatel-Lucent Architecture and Integration Via C.G. Viola 65 00148-Rome-Italy Email:mailto:esampaolesi at alcatel-lucent.com Phone: (+39) 0665182708 Mobile: (+39) 3482889759 Fax: (+39) 0665182104 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://sipforum.org/pipermail/discussion/attachments/20071030/689871d7/attachment.html From deveshbissa at rediffmail.com Mon Oct 1 08:15:13 2007 From: deveshbissa at rediffmail.com (devesh bissa) Date: 1 Oct 2007 08:15:13 -0000 Subject: [SIPForum-discussion] border controller for sip Message-ID: <20071001081513.32247.qmail@f5mail-237-201.rediffmail.com> Hi, I want to use border controller(openSBC) with IMS network for sip.Please help (how to configure and use it) if anyone did it previously.Thank youDevesh -------------- next part -------------- An HTML attachment was scrubbed... URL: From kit_del_rosario at yahoo.com Mon Oct 1 08:54:52 2007 From: kit_del_rosario at yahoo.com (Francisco del rosario) Date: Mon, 1 Oct 2007 01:54:52 -0700 (PDT) Subject: [SIPForum-discussion] ASTERISK PBX INTEGRATION WITH A SOFTSWITCH Message-ID: <923711.18892.qm@web52011.mail.re2.yahoo.com> Hi, Need your input to get the right script for Asterisk PBX. The issue is for outgoing calls when connected to a mobile device. The call is connected but there is no speech path i.e. no voice received from both ends. Any advice to provide the right script file . ? Thanks... ____________________________________________________________________________________ Moody friends. Drama queens. Your life? Nope! - their life, your story. Play Sims Stories at Yahoo! Games. http://sims.yahoo.com/ -------------- next part -------------- An HTML attachment was scrubbed... URL: From j.scholz at teles.de Mon Oct 1 16:58:09 2007 From: j.scholz at teles.de (Joerg Scholz) Date: Mon, 1 Oct 2007 18:58:09 +0200 Subject: [SIPForum-discussion] specific case with a=inactive Message-ID: Hello everybody, I have a quick question related to the SDP signaling behavior in a specific hold case. My scenario is the following: Incoming Re-invite of an established voice call with the following content of SDP: c=IN IP4 0.0.0.0 ... a=inactive In the 200ok I confirm the attribute line: a=inactive and the call is hold - everything ok. Later on comes another Re-invite without SDP for the same call. Now my question, if I answer in the 200ok for that Re-invite with SDP do I have to repeat also the: a=inactive attribute or do I send now my SDP again with a=sendrecv Thanks and Best regards Joerg -------------- next part -------------- An HTML attachment was scrubbed... URL: From vivian_cyn at hotmail.com Mon Oct 1 17:03:01 2007 From: vivian_cyn at hotmail.com (ChenVivian) Date: Mon, 1 Oct 2007 17:03:01 +0000 Subject: [SIPForum-discussion] 3pcc implementation with sip provider Message-ID: Hi, My question might be very simple since I am new to SIP. The problem is that I have implemented a third part call controller following flow IV in rfc3725. I tested it using P2P module, it works fine. But when I applied it to sip server, sip.voipstunt.com, which kept sending me the 4XX message (bad request or unsupported media type) on response to my INVITE(no SDP). I don't know what could cause the failure. I was wondering if it is because the sip server doesn't support 3pcc messages. should both sip server and gateway support 3pcc in this situation? Thanks very much. Best Regards,

Yuening Chen

Department of Computer Science,

Uppsala University,

Sweden.

vivian_cyn at hotmail.com

_________________________________________________________________ Windows Live Spaces ???????? http://miaomiaogarden2007.spaces.live.com/ -------------- next part -------------- An HTML attachment was scrubbed... URL: From rjsparks at nostrum.com Mon Oct 1 20:26:45 2007 From: rjsparks at nostrum.com (Robert Sparks) Date: Mon, 1 Oct 2007 15:26:45 -0500 Subject: [SIPForum-discussion] SIPit 21 registration closes Oct 29 Message-ID: <1EC7DF16-E27B-4E6F-AEB5-DEA4C284E7C4@nostrum.com> SIPit 21 will take place November 5 through 9, 2007 in Beijing, China. Registration will close October 29 (four weeks from now). If you haven't already registered, please reserve your seat now before the event fills up. This SIPit will be hosted by the BII Group and the Beijing University of Posts and Telecommunications. The registration fee is $575 US Dollars per participant. See http://www.sipit.net for more information and to register. See you in Beijing! RjS From robert.traussnig at kapsch.net Tue Oct 2 05:20:34 2007 From: robert.traussnig at kapsch.net (Traussnig Robert) Date: Tue, 2 Oct 2007 07:20:34 +0200 Subject: [SIPForum-discussion] specific case with a=inactive In-Reply-To: Message-ID: <23D52CF27A66BF4E9352313D60E46F3E0113EA3D@EXCLUSTER.kcc.local> Hi! I'm not quite sure if this is correct but because of the fact that you didn't get a SDP in the Re-Invite you have to give an offer in your 200ok. So you have to decide if you send a=inactive again or a=sendrecv. Maybe someone have a better answer. Best Regards, Robert -----Ursprungliche Nachricht----- Von: discussion-bounces at sipforum.org [mailto:discussion-bounces at sipforum.org]Im Auftrag von Joerg Scholz Gesendet: Montag, 01. Oktober 2007 18:58 An: discussion at sipforum.org Betreff: [SIPForum-discussion] specific case with a=inactive Hello everybody, I have a quick question related to the SDP signaling behavior in a specific hold case. My scenario is the following: Incoming Re-invite of an established voice call with the following content of SDP: c=IN IP4 0.0.0.0 ... a=inactive In the 200ok I confirm the attribute line: a=inactive and the call is hold - everything ok. Later on comes another Re-invite without SDP for the same call. Now my question, if I answer in the 200ok for that Re-invite with SDP do I have to repeat also the: a=inactive attribute or do I send now my SDP again with a=sendrecv Thanks and Best regards Joerg -------------- next part -------------- An HTML attachment was scrubbed... URL: From kunusan at yahoo.com Tue Oct 2 08:40:57 2007 From: kunusan at yahoo.com (badal naik) Date: Tue, 2 Oct 2007 01:40:57 -0700 (PDT) Subject: [SIPForum-discussion] specific case with a=inactive In-Reply-To: <23D52CF27A66BF4E9352313D60E46F3E0113EA3D@EXCLUSTER.kcc.local> Message-ID: <288012.81621.qm@web56704.mail.re3.yahoo.com> Well, I think in your scenario U have to add the SDP parameter of the previous success(2**) reponse. Re-Invite should work as follows in ur case: When a UAC sends a re-invite with no session description, in which case the first reliable non-failure response to the re-invite will contain the offer. That is my understanding.. Thanks Badal Naik --- Traussnig Robert wrote: > Hi! > > I'm not quite sure if this is correct but because of > the fact that you > didn't get a SDP in the Re-Invite you have to give > an offer in your > 200ok. So you have to decide if you send a=inactive > again or a=sendrecv. > Maybe someone have a better answer. > > Best Regards, > Robert > > > -----Ursprungliche Nachricht----- > Von: discussion-bounces at sipforum.org > [mailto:discussion-bounces at sipforum.org]Im Auftrag > von Joerg Scholz > Gesendet: Montag, 01. Oktober 2007 18:58 > An: discussion at sipforum.org > Betreff: [SIPForum-discussion] specific case with > a=inactive > > > > Hello everybody, > > I have a quick question related to the SDP signaling > behavior in a > specific hold case. My scenario is the following: > > > > Incoming Re-invite of an established voice call with > the following > content of SDP: > > c=IN IP4 0.0.0.0 > > ... > > a=inactive > > > > In the 200ok I confirm the attribute line: > > a=inactive > > > > and the call is hold - everything ok. > > Later on comes another Re-invite without SDP for the > same call. Now my > question, if I answer in the 200ok for that > Re-invite with SDP do I have > to repeat also the: > > a=inactive > > attribute or do I send now my SDP again with > > a=sendrecv > > > > Thanks and Best regards > > Joerg > > > > > _______________________________________________ > This is the SIP Forum discussion mailing list > TO UNSUBSCRIBE, or edit your delivery options, > please visit > http://sipforum.org/mailman/listinfo/discussion > Post to the list at discussion at sipforum.org > ____________________________________________________________________________________ Yahoo! oneSearch: Finally, mobile search that gives answers, not web links. http://mobile.yahoo.com/mobileweb/onesearch?refer=1ONXIC From j.scholz at teles.de Tue Oct 2 09:05:08 2007 From: j.scholz at teles.de (Joerg Scholz) Date: Tue, 2 Oct 2007 11:05:08 +0200 Subject: [SIPForum-discussion] specific case with a=inactive Message-ID: Thanks Badel and Robert for your responds. That's also what I expected to do - but it seems to be a problem for that class 5 SIP server solution which I try to work with. It uses the Re-invite without SDP to retrieve the hold call. Thereby the retrieve is initiated with a user agent web application (not the phone). If I send still: A=inactive In the SDP of the 200ok response of my device; it is used for the second leg of the call as SDP content and the call will be reestablished without voice. So I need exactly to know what the right behavior is in that case. It is quite simple to change the behavior but I'm pretty sure that other cases will have problem then. Thanks again and best regards Joerg -----Original Message----- From: badal naik [mailto:kunusan at yahoo.com] Sent: Tuesday, October 02, 2007 10:41 AM To: Traussnig Robert; Joerg Scholz; discussion at sipforum.org Subject: Re: [SIPForum-discussion] specific case with a=inactive Well, I think in your scenario U have to add the SDP parameter of the previous success(2**) reponse. Re-Invite should work as follows in ur case: When a UAC sends a re-invite with no session description, in which case the first reliable non-failure response to the re-invite will contain the offer. That is my understanding.. Thanks Badal Naik --- Traussnig Robert wrote: > Hi! > > I'm not quite sure if this is correct but because of > the fact that you > didn't get a SDP in the Re-Invite you have to give > an offer in your > 200ok. So you have to decide if you send a=inactive > again or a=sendrecv. > Maybe someone have a better answer. > > Best Regards, > Robert > > > -----Ursprungliche Nachricht----- > Von: discussion-bounces at sipforum.org > [mailto:discussion-bounces at sipforum.org]Im Auftrag > von Joerg Scholz > Gesendet: Montag, 01. Oktober 2007 18:58 > An: discussion at sipforum.org > Betreff: [SIPForum-discussion] specific case with > a=inactive > > > > Hello everybody, > > I have a quick question related to the SDP signaling > behavior in a > specific hold case. My scenario is the following: > > > > Incoming Re-invite of an established voice call with > the following > content of SDP: > > c=IN IP4 0.0.0.0 > > ... > > a=inactive > > > > In the 200ok I confirm the attribute line: > > a=inactive > > > > and the call is hold - everything ok. > > Later on comes another Re-invite without SDP for the > same call. Now my > question, if I answer in the 200ok for that > Re-invite with SDP do I have > to repeat also the: > > a=inactive > > attribute or do I send now my SDP again with > > a=sendrecv > > > > Thanks and Best regards > > Joerg > > > > > _______________________________________________ > This is the SIP Forum discussion mailing list > TO UNSUBSCRIBE, or edit your delivery options, > please visit > http://sipforum.org/mailman/listinfo/discussion > Post to the list at discussion at sipforum.org > ____________________________________________________________________________ ________ Yahoo! oneSearch: Finally, mobile search that gives answers, not web links. http://mobile.yahoo.com/mobileweb/onesearch?refer=1ONXIC -------------- next part -------------- An HTML attachment was scrubbed... URL: From kunusan at yahoo.com Tue Oct 2 11:10:25 2007 From: kunusan at yahoo.com (badal naik) Date: Tue, 2 Oct 2007 04:10:25 -0700 (PDT) Subject: [SIPForum-discussion] specific case with a=inactive In-Reply-To: Message-ID: <858677.33596.qm@web56704.mail.re3.yahoo.com> Hello Joerg, I am not sure whether this will be helpful to you.But take it as a suggestion. As i Know call hold is implemented in two ways. 1. Send a re-Invite with "0.0.0.0" as the IP address in the sdp data.(RFC 2543) 2.Send a re-Invite with the parameter a=sendonly set in the sdp data.(RFC 3264 ) You have tried the first one.Can u check the authenticity of the second one. In the first case the recepient cant send RTCP to you.I think most advanced server uses RFC 3264 for call Hold.That may be the case that ur server expects this RFC and you are following the obsolete one. Try with second one.I think it may work.. But please let me know if things work out. Thanks Badal Naik --- Joerg Scholz wrote: > Thanks Badel and Robert > for your responds. That's also what I expected to do > - but it seems to be a > problem for that class 5 SIP server solution which I > try to work with. > It uses the Re-invite without SDP to retrieve the > hold call. Thereby the > retrieve is initiated with a user agent web > application (not the phone). If > I send still: > A=inactive > In the SDP of the 200ok response of my device; it is > used for the second leg > of the call as SDP content and the call will be > reestablished without voice. > > So I need exactly to know what the right behavior is > in that case. > > It is quite simple to change the behavior but I'm > pretty sure that other > cases will have problem then. > Thanks again and best regards > Joerg > > > -----Original Message----- > From: badal naik [mailto:kunusan at yahoo.com] > Sent: Tuesday, October 02, 2007 10:41 AM > To: Traussnig Robert; Joerg Scholz; > discussion at sipforum.org > Subject: Re: [SIPForum-discussion] specific case > with a=inactive > > Well, > I think in your scenario U have to add the SDP > parameter of the previous success(2**) reponse. > Re-Invite should work as follows in ur case: > When a UAC sends a re-invite with no session > description, in which case the first reliable > non-failure response to the re-invite will contain > the > offer. > > That is my understanding.. > > Thanks > Badal Naik > --- Traussnig Robert > wrote: > > > Hi! > > > > I'm not quite sure if this is correct but because > of > > the fact that you > > didn't get a SDP in the Re-Invite you have to give > > an offer in your > > 200ok. So you have to decide if you send > a=inactive > > again or a=sendrecv. > > Maybe someone have a better answer. > > > > Best Regards, > > Robert > > > > > > -----Ursprungliche Nachricht----- > > Von: discussion-bounces at sipforum.org > > [mailto:discussion-bounces at sipforum.org]Im Auftrag > > von Joerg Scholz > > Gesendet: Montag, 01. Oktober 2007 18:58 > > An: discussion at sipforum.org > > Betreff: [SIPForum-discussion] specific case with > > a=inactive > > > > > > > > Hello everybody, > > > > I have a quick question related to the SDP > signaling > > behavior in a > > specific hold case. My scenario is the following: > > > > > > > > Incoming Re-invite of an established voice call > with > > the following > > content of SDP: > > > > c=IN IP4 0.0.0.0 > > > > ... > > > > a=inactive > > > > > > > > In the 200ok I confirm the attribute line: > > > > a=inactive > > > > > > > > and the call is hold - everything ok. > > > > Later on comes another Re-invite without SDP for > the > > same call. Now my > > question, if I answer in the 200ok for that > > Re-invite with SDP do I have > > to repeat also the: > > > > a=inactive > > > > attribute or do I send now my SDP again with > > > > a=sendrecv > > > > > > > > Thanks and Best regards > > > > Joerg > > > > > > > > > _______________________________________________ > > This is the SIP Forum discussion mailing list > > TO UNSUBSCRIBE, or edit your delivery options, > > please visit > > http://sipforum.org/mailman/listinfo/discussion > > Post to the list at discussion at sipforum.org > > > > > > > ____________________________________________________________________________ > ________ > Yahoo! oneSearch: Finally, mobile search > that gives answers, not web links. > http://mobile.yahoo.com/mobileweb/onesearch?refer=1ONXIC > ____________________________________________________________________________________ Check out the hottest 2008 models today at Yahoo! Autos. http://autos.yahoo.com/new_cars.html From victor.pascual.avila at gmail.com Tue Oct 2 12:23:15 2007 From: victor.pascual.avila at gmail.com (=?ISO-8859-1?Q?Victor_Pascual_=C1vila?=) Date: Tue, 2 Oct 2007 14:23:15 +0200 Subject: [SIPForum-discussion] border controller for sip In-Reply-To: <20071001081513.32247.qmail@f5mail-237-201.rediffmail.com> References: <20071001081513.32247.qmail@f5mail-237-201.rediffmail.com> Message-ID: <618e24240710020523m2896f194l9b4ba8c0159f4c54@mail.gmail.com> Hello, check the following link. http://www.opensipstack.org/sbc_man_quickstart.html I hope it'll be useful, Victor Pascual On 1 Oct 2007 08:15:13 -0000, devesh bissa wrote: > > Hi, > I want to use border controller(openSBC) with IMS network for sip. > Please help (how to configure and use it) if anyone did it previously. > > Thank you > Devesh > [image: sig js] > _______________________________________________ > This is the SIP Forum discussion mailing list > TO UNSUBSCRIBE, or edit your delivery options, please visit > http://sipforum.org/mailman/listinfo/discussion > Post to the list at discussion at sipforum.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From meharchaoui28 at googlemail.com Tue Oct 2 12:47:58 2007 From: meharchaoui28 at googlemail.com (mohammed El harchaoui) Date: Tue, 2 Oct 2007 14:47:58 +0200 Subject: [SIPForum-discussion] CSDM problem Message-ID: <6f011d50710020547k64813c97ofb8873da7185d7f5@mail.gmail.com> Hi all, I'm trying to develop a sip mobile client, that uses xcap to manage resource list. so if the client needs for example to modify a resource list, he MUST send a "PUT" http message to the csdm server, but the problem is that the mobile devices on which the client should be run do not support the mentioned method above(PUT and also DELETE). DOes anyone solved this problem before, plzzzz help me!!! Mohammed -------------- next part -------------- An HTML attachment was scrubbed... URL: From PScheffler at carrieraccess.com Tue Oct 2 13:59:11 2007 From: PScheffler at carrieraccess.com (Scheffler, Paul) Date: Tue, 2 Oct 2007 07:59:11 -0600 Subject: [SIPForum-discussion] specific case with a=inactive In-Reply-To: Message-ID: <33E402324D746F48AF9FBAC491FF5C8C4CE6C1@camailsvr01.carrieraccess.com> Hello Joerg: My recommendation (when you get a re-INVITE with no SDP) is to ignore the current call state, and send an SDP response in the 200 OK which represents your normal offer when initiating a new call. There are some SIP application servers which expect this, because they are trying to determine your normal capabilities in preparation for reconnecting your call. If you do this, you should not have audio problems. Paul Scheffler Carrier Access Corp. Boulder, CO ________________________________ From: discussion-bounces at sipforum.org [mailto:discussion-bounces at sipforum.org] On Behalf Of Joerg Scholz Sent: Tuesday, October 02, 2007 3:05 AM To: 'badal naik'; Traussnig Robert; discussion at sipforum.org Subject: Re: [SIPForum-discussion] specific case with a=inactive Thanks Badel and Robert for your responds. That's also what I expected to do - but it seems to be a problem for that class 5 SIP server solution which I try to work with. It uses the Re-invite without SDP to retrieve the hold call. Thereby the retrieve is initiated with a user agent web application (not the phone). If I send still: A=inactive In the SDP of the 200ok response of my device; it is used for the second leg of the call as SDP content and the call will be reestablished without voice. So I need exactly to know what the right behavior is in that case. It is quite simple to change the behavior but I'm pretty sure that other cases will have problem then. Thanks again and best regards Joerg -----Original Message----- From: badal naik [mailto:kunusan at yahoo.com] Sent: Tuesday, October 02, 2007 10:41 AM To: Traussnig Robert; Joerg Scholz; discussion at sipforum.org Subject: Re: [SIPForum-discussion] specific case with a=inactive Well, I think in your scenario U have to add the SDP parameter of the previous success(2**) reponse. Re-Invite should work as follows in ur case: When a UAC sends a re-invite with no session description, in which case the first reliable non-failure response to the re-invite will contain the offer. That is my understanding.. Thanks Badal Naik --- Traussnig Robert wrote: > Hi! > > I'm not quite sure if this is correct but because of > the fact that you > didn't get a SDP in the Re-Invite you have to give > an offer in your > 200ok. So you have to decide if you send a=inactive > again or a=sendrecv. > Maybe someone have a better answer. > > Best Regards, > Robert > > > -----Ursprungliche Nachricht----- > Von: discussion-bounces at sipforum.org > [mailto:discussion-bounces at sipforum.org]Im Auftrag > von Joerg Scholz > Gesendet: Montag, 01. Oktober 2007 18:58 > An: discussion at sipforum.org > Betreff: [SIPForum-discussion] specific case with > a=inactive > > > > Hello everybody, > > I have a quick question related to the SDP signaling > behavior in a > specific hold case. My scenario is the following: > > > > Incoming Re-invite of an established voice call with > the following > content of SDP: > > c=IN IP4 0.0.0.0 > > ... > > a=inactive > > > > In the 200ok I confirm the attribute line: > > a=inactive > > > > and the call is hold - everything ok. > > Later on comes another Re-invite without SDP for the > same call. Now my > question, if I answer in the 200ok for that > Re-invite with SDP do I have > to repeat also the: > > a=inactive > > attribute or do I send now my SDP again with > > a=sendrecv > > > > Thanks and Best regards > > Joerg > > > > > _______________________________________________ > This is the SIP Forum discussion mailing list > TO UNSUBSCRIBE, or edit your delivery options, > please visit > http://sipforum.org/mailman/listinfo/discussion > Post to the list at discussion at sipforum.org > ________________________________________________________________________ ____________ Yahoo! oneSearch: Finally, mobile search that gives answers, not web links. http://mobile.yahoo.com/mobileweb/onesearch?refer=1ONXIC -------------- next part -------------- An HTML attachment was scrubbed... URL: From j.scholz at teles.de Tue Oct 2 14:22:00 2007 From: j.scholz at teles.de (Joerg Scholz) Date: Tue, 2 Oct 2007 16:22:00 +0200 Subject: [SIPForum-discussion] specific case with a=inactive Message-ID: Dear Paul, Thank you; I will do so and ignore for now other optional scenarios which might have a problem with that implementation. As I understand it at the moment is there no clear solution for that. Best regards Joerg -----Original Message----- From: Scheffler, Paul [mailto:PScheffler at carrieraccess.com] Sent: Tuesday, October 02, 2007 3:59 PM To: Joerg Scholz; badal naik; Traussnig Robert; discussion at sipforum.org Subject: RE: [SIPForum-discussion] specific case with a=inactive Hello Joerg: My recommendation (when you get a re-INVITE with no SDP) is to ignore the current call state, and send an SDP response in the 200 OK which represents your normal offer when initiating a new call. There are some SIP application servers which expect this, because they are trying to determine your normal capabilities in preparation for reconnecting your call. If you do this, you should not have audio problems. Paul Scheffler Carrier Access Corp. Boulder, CO _____ From: discussion-bounces at sipforum.org [mailto:discussion-bounces at sipforum.org] On Behalf Of Joerg Scholz Sent: Tuesday, October 02, 2007 3:05 AM To: 'badal naik'; Traussnig Robert; discussion at sipforum.org Subject: Re: [SIPForum-discussion] specific case with a=inactive Thanks Badel and Robert for your responds. That's also what I expected to do - but it seems to be a problem for that class 5 SIP server solution which I try to work with. It uses the Re-invite without SDP to retrieve the hold call. Thereby the retrieve is initiated with a user agent web application (not the phone). If I send still: A=inactive In the SDP of the 200ok response of my device; it is used for the second leg of the call as SDP content and the call will be reestablished without voice. So I need exactly to know what the right behavior is in that case. It is quite simple to change the behavior but I'm pretty sure that other cases will have problem then. Thanks again and best regards Joerg -----Original Message----- From: badal naik [mailto:kunusan at yahoo.com ] Sent: Tuesday, October 02, 2007 10:41 AM To: Traussnig Robert; Joerg Scholz; discussion at sipforum.org Subject: Re: [SIPForum-discussion] specific case with a=inactive Well, I think in your scenario U have to add the SDP parameter of the previous success(2**) reponse. Re-Invite should work as follows in ur case: When a UAC sends a re-invite with no session description, in which case the first reliable non-failure response to the re-invite will contain the offer. That is my understanding.. Thanks Badal Naik --- Traussnig Robert wrote: > Hi! > > I'm not quite sure if this is correct but because of > the fact that you > didn't get a SDP in the Re-Invite you have to give > an offer in your > 200ok. So you have to decide if you send a=inactive > again or a=sendrecv. > Maybe someone have a better answer. > > Best Regards, > Robert > > > -----Ursprungliche Nachricht----- > Von: discussion-bounces at sipforum.org > [mailto:discussion-bounces at sipforum.org ]Im Auftrag > von Joerg Scholz > Gesendet: Montag, 01. Oktober 2007 18:58 > An: discussion at sipforum.org > Betreff: [SIPForum-discussion] specific case with > a=inactive > > > > Hello everybody, > > I have a quick question related to the SDP signaling > behavior in a > specific hold case. My scenario is the following: > > > > Incoming Re-invite of an established voice call with > the following > content of SDP: > > c=IN IP4 0.0.0.0 > > ... > > a=inactive > > > > In the 200ok I confirm the attribute line: > > a=inactive > > > > and the call is hold - everything ok. > > Later on comes another Re-invite without SDP for the > same call. Now my > question, if I answer in the 200ok for that > Re-invite with SDP do I have > to repeat also the: > > a=inactive > > attribute or do I send now my SDP again with > > a=sendrecv > > > > Thanks and Best regards > > Joerg > > > > > _______________________________________________ > This is the SIP Forum discussion mailing list > TO UNSUBSCRIBE, or edit your delivery options, > please visit > http://sipforum.org/mailman/listinfo/discussion > Post to the list at discussion at sipforum.org > ____________________________________________________________________________ ________ Yahoo! oneSearch: Finally, mobile search that gives answers, not web links. http://mobile.yahoo.com/mobileweb/onesearch?refer=1ONXIC -------------- next part -------------- An HTML attachment was scrubbed... URL: From zmihaly at madein.hu Tue Oct 2 16:07:10 2007 From: zmihaly at madein.hu (Mihaly Zachar) Date: Tue, 02 Oct 2007 18:07:10 +0200 Subject: [SIPForum-discussion] 302 message question Message-ID: <47026CAE.6030603@madein.hu> Hi all, I'm writing an UAS. The UAS has a feature, that if the called number is matching with a pattern, it will send 183 Session in progress, than plays an RTP stream and then redirect the UAC with 302 Moved Temporarily.. There is an UAC, and it's developers says that I should not send 302 Redirect after the 183 Session in progress. This solution works well with CISCO media gateways. I can't find it in the RFC 3261 who has the truth.. Can sy help me in this ? So, the call flow is the following: UAC UAS --- INVITE ---> <--- 100 ------ <--- 183 ------ <-- RTP -- . . . <--- 302 ---- Is this correct ? Thanks, Misi From sreekant_nair at yahoo.com Tue Oct 2 18:05:42 2007 From: sreekant_nair at yahoo.com (sreekant nair) Date: Tue, 2 Oct 2007 11:05:42 -0700 (PDT) Subject: [SIPForum-discussion] 302 message question Message-ID: <11724.69746.qm@web51107.mail.re2.yahoo.com> Check out this link. http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/rel_docs/sip_flo/hennigan.htm There is a call flow depicting the messaging for a scenario where a 3XX response is received after a 183 is sent by the server. I guess that explains how CISCO supports it. But yeah I need to dig deeper to find an RFC that states this. Regards Sreekant ----- Original Message ---- From: Mihaly Zachar To: discussion at sipforum.org Sent: Tuesday, October 2, 2007 12:07:10 PM Subject: [SIPForum-discussion] 302 message question Hi all, I'm writing an UAS. The UAS has a feature, that if the called number is matching with a pattern, it will send 183 Session in progress, than plays an RTP stream and then redirect the UAC with 302 Moved Temporarily.. There is an UAC, and it's developers says that I should not send 302 Redirect after the 183 Session in progress. This solution works well with CISCO media gateways. I can't find it in the RFC 3261 who has the truth.. Can sy help me in this ? So, the call flow is the following: UAC UAS --- INVITE ---> <--- 100 ------ <--- 183 ------ <-- RTP -- . . . <--- 302 ---- Is this correct ? Thanks, Misi _______________________________________________ This is the SIP Forum discussion mailing list TO UNSUBSCRIBE, or edit your delivery options, please visit http://sipforum.org/mailman/listinfo/discussion Post to the list at discussion at sipforum.org ____________________________________________________________________________________ Looking for a deal? Find great prices on flights and hotels with Yahoo! FareChase. http://farechase.yahoo.com/ -------------- next part -------------- An HTML attachment was scrubbed... URL: From amos.halfon at gmail.com Wed Oct 3 05:49:20 2007 From: amos.halfon at gmail.com (Amos Halfon) Date: Wed, 3 Oct 2007 07:49:20 +0200 Subject: [SIPForum-discussion] (no subject) Message-ID: <697963e10710022249o2bd13e6ayd95a4d6c12a97794@mail.gmail.com> -------------- next part -------------- An HTML attachment was scrubbed... URL: From louis at ttv.com.hk Wed Oct 3 06:11:58 2007 From: louis at ttv.com.hk (Louis Wu) Date: Wed, 3 Oct 2007 14:11:58 +0800 Subject: [SIPForum-discussion] Call Disconnect issue with Cisco AS5300 running SIP Message-ID: <20071003055430.M56996@ttv.com.hk> Hi All, I have a Cisco AS5300 using SIP and initiate SIP call to a SIP server YATE (v 1.3.0). I have a call disconnect problem whenever my Cisco receive a 183 Session Progress message from the YATE server. The symptoms are listed as below. 1. Cisco AS5300 send an INVITE to the YATE server 2. YATE returns a 100 Trying message 3. YATE returns a 183 Session Progress 4. YATE returns a 200 OK 5. Two-way-audio starts (start conversation as usual), but at the ISDN side of the Cisco, the call is shown to be "not connected" 6. Cisco sends a ACK 7. Cisco sends a BYE 8. YATE returns a 100 Trying 9. Cisco sends a BYE 10. YATE returns a 200 OK 11. Call disconnect with status message saying "no answer" at the calling party's mobile handset 12. Cisco logs a Disconnect Cause (CC) : 16 (SIP) : 200 If the YATE returns a 180 Session Progress in (3) above, the call will be connected normally and works as usual. Please give me your professional advice and resolution on the above disconnect issue. Cheers Louis From mhiqe at yahoo.com Wed Oct 3 08:30:41 2007 From: mhiqe at yahoo.com (Mhike) Date: Wed, 3 Oct 2007 01:30:41 -0700 (PDT) Subject: [SIPForum-discussion] FMTP / RTPMAP Message-ID: <486570.72622.qm@web50811.mail.re2.yahoo.com> Hi, Can anyone tell me what's the meaning of FMTP and RTPMAP? What are their differences? Thanks. ____________________________________________________________________________________ Don't let your dream ride pass you by. Make it a reality with Yahoo! Autos. http://autos.yahoo.com/index.html -------------- next part -------------- An HTML attachment was scrubbed... URL: From abhishek.mishra at globallogic.com Wed Oct 3 08:47:18 2007 From: abhishek.mishra at globallogic.com (Abhishek Mishra) Date: Wed, 03 Oct 2007 14:17:18 +0530 Subject: [SIPForum-discussion] FMTP / RTPMAP In-Reply-To: <486570.72622.qm@web50811.mail.re2.yahoo.com> References: <486570.72622.qm@web50811.mail.re2.yahoo.com> Message-ID: <1191401237.2587.6.camel@linux.site> Hi Mhike, Please refer to RFC 2327 and RFC 3264: http://tools.ietf.org/html/rfc2327 Kind Regards, -Abhishek On Wed, 2007-10-03 at 14:00, Mhike wrote: > Hi, > > Can anyone tell me what's the meaning of FMTP and RTPMAP? What are > their differences? > > Thanks. > > > ______________________________________________________________________ > Fussy? Opinionated? Impossible to please? Perfect. Join Yahoo!'s user > panel and lay it on us. > > ______________________________________________________________________ > _______________________________________________ > This is the SIP Forum discussion mailing list > TO UNSUBSCRIBE, or edit your delivery options, please visit http://sipforum.org/mailman/listinfo/discussion > Post to the list at discussion at sipforum.org > From ian.sivell at gmail.com Wed Oct 3 11:36:29 2007 From: ian.sivell at gmail.com (Ian Sivell) Date: Wed, 3 Oct 2007 12:36:29 +0100 Subject: [SIPForum-discussion] CISCO 7940 TFTP Timeout Message-ID: <123aa00e0710030436y6996fd33o577aed42e77b49f@mail.gmail.com> Hi, I hope someone can help me I have recently bought a used 7940 from EBay, initially it booted and I could get into the menus (all be them locked), I found on the cisco site to hold down th # key whilst powering up and then enter 123456789*0# to reset toi factory defaults. Since doing is the phone boots but just stays at the tftp timeout message. On the Cisco site it says that the phone should timeout after 20 seconds and continue to boot correctly after that giving access to the menus mine does not seem to do this. It has been loaded with SCCP (Skinny) as far as I can tell, and the DHCP server on my network is assigning it a DHCP address within a valid subnet but still it times out and does not get any further than the message above. I have reset this several times to n avail Please some one help me Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: From yasin at kaplan.net Wed Oct 3 11:46:47 2007 From: yasin at kaplan.net (Yasin KAPLAN) Date: Wed, 3 Oct 2007 14:46:47 +0300 Subject: [SIPForum-discussion] TekSIP Message-ID: <015001c805b3$163e84e0$1a0d3ad4@doruk.com.tr> Hi, I've recently released beta version of TekSIP Registrar & Proxy: http://www.teksip.com/ You feedback is welcomed. Thanks, Yasin KAPLAN -------------- next part -------------- An HTML attachment was scrubbed... URL: From amit.v at pyronetworks.com Wed Oct 3 13:46:03 2007 From: amit.v at pyronetworks.com (amit) Date: Wed, 03 Oct 2007 19:16:03 +0530 Subject: [SIPForum-discussion] Image with invite msg Message-ID: <1191419164.5119.3.camel@amit> Hi All, How we send image with sip invite message ? Thanks in Advance Amit From mohamed2005777 at yahoo.com Wed Oct 3 14:41:03 2007 From: mohamed2005777 at yahoo.com (mohamed hamdy) Date: Wed, 3 Oct 2007 07:41:03 -0700 (PDT) Subject: [SIPForum-discussion] Fwd: Call Disconnect issue with Cisco AS5300 running SIP Message-ID: <963352.71849.qm@web56404.mail.re3.yahoo.com> Hi,every one I'm now working for voip call signalling over sip and i want really a support to download cisco sip proxy server for free ( without paying) plz help me as soon as possible becouse the end of the project will be within month Note: forwarded message attached. --------------------------------- Building a website is a piece of cake. Yahoo! Small Business gives you all the tools to get online. -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- An embedded message was scrubbed... From: "Louis Wu" Subject: [SIPForum-discussion] Call Disconnect issue with Cisco AS5300 running SIP Date: Wed, 3 Oct 2007 14:11:58 +0800 Size: 3918 URL: From raymond.jender.ctr at disa.mil Wed Oct 3 15:28:10 2007 From: raymond.jender.ctr at disa.mil (Jender, Raymond C CTR DISA JITC) Date: Wed, 3 Oct 2007 08:28:10 -0700 Subject: [SIPForum-discussion] Fwd: Call Disconnect issue with Cisco AS5300running SIP (UNCLASSIFIED) In-Reply-To: <963352.71849.qm@web56404.mail.re3.yahoo.com> References: <963352.71849.qm@web56404.mail.re3.yahoo.com> Message-ID: <00E9117C300386479B7E2F3CDBF798F601E7EC5F@pothia.disanet.disa-u.mil> Classification: UNCLASSIFIED Caveats: NONE If the Cisco SIP Proxy Server is a commercial product, you should buy it and not look for bootlegged copies. Otherwie, there are other free sip proxies out there.... Raymond C. Jender Booz|Allen|Hamilton DSN IA Test Team Ft. Huachuca, Az. 520-538-2588 -----Original Message----- From: discussion-bounces at sipforum.org [mailto:discussion-bounces at sipforum.org] On Behalf Of mohamed hamdy Sent: Wednesday, October 03, 2007 7:41 AM To: discussion at sipforum.org Subject: [SIPForum-discussion] Fwd: Call Disconnect issue with Cisco AS5300running SIP Hi,every one I'm now working for voip call signalling over sip and i want really a support to download cisco sip proxy server for free ( without paying) plz help me as soon as possible becouse the end of the project will be within month Note: forwarded message attached. ________________________________ Building a website is a piece of cake. Yahoo! Small Business gives you all the tools to get online. Classification: UNCLASSIFIED Caveats: NONE From gpaul at aylus.com Wed Oct 3 15:43:11 2007 From: gpaul at aylus.com (Geo Paul) Date: Wed, 3 Oct 2007 11:43:11 -0400 Subject: [SIPForum-discussion] FMTP / RTPMAP In-Reply-To: <486570.72622.qm@web50811.mail.re2.yahoo.com> Message-ID: When ever a dynamic payload type is used in the sdp or when ever additional information is required to decode, the additional information should be given in a=rtpmap: /[/] a=fmtp: This attribute allows parameters that are specific to a particular format to be conveyed in a way that SDP doesn't have to understand them. The format must be one of the formats specified for the media. Format-specific parameters may be any set of parameters required to be conveyed by SDP and given unchanged to the media tool that will use this format. It is a media attribute, and is not dependent on charset. Geo _____ From: Mhike [mailto:mhiqe at yahoo.com] Sent: Wednesday, October 03, 2007 4:31 AM To: discussion at sipforum.org Subject: [SIPForum-discussion] FMTP / RTPMAP Hi, Can anyone tell me what's the meaning of FMTP and RTPMAP? What are their differences? Thanks. _____ Fussy? Opinionated? Impossible to please? Perfect. Join Yahoo!'s user panel and lay it on us. -------------- next part -------------- An HTML attachment was scrubbed... URL: From cross at gocross.com Wed Oct 3 16:30:26 2007 From: cross at gocross.com (Tom Cross) Date: Wed, 3 Oct 2007 10:30:26 -0600 Subject: [SIPForum-discussion] Book Review of "Securing VoIP Networks" Message-ID: <014e01c805da$b6bedbf0$679c0818@dv5020us> Please pass along. "Once in a blue moon you read a book that not just meets but beats your expectations. The book I am referring to is Securing VoIP Networks by Peter Thermos and Ari Takanen, Addison-Wesley, ISBN-0-321-43734-9, www.awprofessional.com . The more you read, the more you want to read. All too often technical books are too-deep or too-high. This book provides practical, understandable and most importantly, implementable (new word) information. As a SIP course developer and trainer, I am always looking for something to help students learn and do more. This book does that and more. Guess by now you know how I feel, so I will stop." Tom Cross - CEO TECHtionary.com Cheers, Join the CrossTalk blog on TMCnet - http://blog.tmcnet.com/cross-talk/ See all the new Digital Communications, VoIP, SIP and Advanced Network courses at: http://www.techtionary.com - TECHtionary - The World's Largest Animated Library on TECHnology Web Hosting Magazine's Editor's Choice for Technical Help -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.jpg Type: image/jpeg Size: 2743 bytes Desc: not available URL: From chauhan_delhi at yahoo.com Wed Oct 3 16:42:57 2007 From: chauhan_delhi at yahoo.com (Ramesh Chauhan) Date: Wed, 3 Oct 2007 09:42:57 -0700 (PDT) Subject: [SIPForum-discussion] DTMF Issue - Asterisk Message-ID: <309731.93330.qm@web34409.mail.mud.yahoo.com> Hi, I am using Dax IP Phone (Model: DX-301P, H/W Ver:5.1). SIP.CONF : [7711] type=friend username=7711 secret=7711 host=dynamic port=5060 dtmfmode=inband canreinvite=no context=outgoing_ISD mailbox=7711 at default disallow=all allow=ulaw Problem: 7711 is assigned to DAX IP Phone. DTMF is not working. i have tried with already the above settings in SIP.CONF with info, inband, rfc2833, auto. Suggest what to do, so that our DTMF works with this device. Regards Chauhan --------------------------------- Building a website is a piece of cake. Yahoo! Small Business gives you all the tools to get online. -------------- next part -------------- An HTML attachment was scrubbed... URL: From joel.silva at novabase.pt Wed Oct 3 16:49:43 2007 From: joel.silva at novabase.pt (Joel Silva) Date: Wed, 3 Oct 2007 17:49:43 +0100 Subject: [SIPForum-discussion] Presence for multiple publicIDs References: Message-ID: Hello. I was trying to make an application that give me the presence of a number of pubIds associate with a privateId. Imagine that I have associated with my privateId, two publicIDs, one for my sip phone and other to my sip app. I would like that other users could see my name and then a tree associated with the presence of my publicIds. Is this possible? How can I do it? Thanks, Joel. -------------- next part -------------- An HTML attachment was scrubbed... URL: From victor.pascual.avila at gmail.com Wed Oct 3 17:34:34 2007 From: victor.pascual.avila at gmail.com (=?ISO-8859-1?Q?Victor_Pascual_=C1vila?=) Date: Wed, 3 Oct 2007 19:34:34 +0200 Subject: [SIPForum-discussion] TekSIP In-Reply-To: <015001c805b3$163e84e0$1a0d3ad4@doruk.com.tr> References: <015001c805b3$163e84e0$1a0d3ad4@doruk.com.tr> Message-ID: <618e24240710031034ic29d72byabfceaf42480f357@mail.gmail.com> Hello Yasin, good job. Have you tested it with several users (stress test) ? Regards, Victor Pascual On 03/10/2007, Yasin KAPLAN wrote: > > > > > Hi, > > > > I've recently released beta version of TekSIP Registrar & Proxy: > > > > http://www.teksip.com/ > > > > You feedback is welcomed. > > > > Thanks, > > > > Yasin KAPLAN > > > > > _______________________________________________ > This is the SIP Forum discussion mailing list > TO UNSUBSCRIBE, or edit your delivery options, please visit > http://sipforum.org/mailman/listinfo/discussion > Post to the list at discussion at sipforum.org > > From lingyunxjtu at gmail.com Thu Oct 4 06:42:27 2007 From: lingyunxjtu at gmail.com (Karl Tian) Date: Thu, 4 Oct 2007 14:42:27 +0800 Subject: [SIPForum-discussion] The max duration of SIP conversation Message-ID: <807efb400710032342rdb8ca9fyc456c7a3ab622060@mail.gmail.com> Hello everyone, Who can tell me if some rfc protocal(for example: rfc3261) has define the max duration of a sip conversation as 72 hours? Now I'm testing the haleness for a kind of sip client, but the conversations of those clients all stop when the duration reachs to 72 hours. I guess that the question may be caused by a special timer. Please help me about this, thanks! -- Karl.Tian Infinite Shanghai Communication Terminals Ltd. Email :lingyunxjtu at gmail.com Msn:lingyunxjtu at hotmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From VPFR47 at motorola.com Thu Oct 4 06:56:32 2007 From: VPFR47 at motorola.com (S Selvakumar-VPFR47) Date: Thu, 4 Oct 2007 14:56:32 +0800 Subject: [SIPForum-discussion] Asterisk Usage Message-ID: <40E89886C8B3B54B98C5291646C591AA01B6E54A@ZMY16EXM67.ds.mot.com> Hi, I want to know some info about Asterisk. Can anyone help me regards Selva -------------- next part -------------- An HTML attachment was scrubbed... URL: From bn.darshan at gmail.com Thu Oct 4 07:54:36 2007 From: bn.darshan at gmail.com (darshan b n) Date: Thu, 4 Oct 2007 13:24:36 +0530 Subject: [SIPForum-discussion] sips message in ethereal Message-ID: <555d2a660710040054i231a803bua71e70caf2a731ae@mail.gmail.com> Hi all, I want know how to capture a sips message in ethereal? If not is there any other free network protocol analyzer which support this feature? Reply ASAP -- Darshan B N Thanks & Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: From kunusan at yahoo.com Thu Oct 4 08:13:50 2007 From: kunusan at yahoo.com (badal naik) Date: Thu, 4 Oct 2007 01:13:50 -0700 (PDT) Subject: [SIPForum-discussion] The max duration of SIP conversation In-Reply-To: <807efb400710032342rdb8ca9fyc456c7a3ab622060@mail.gmail.com> Message-ID: <200842.58269.qm@web56703.mail.re3.yahoo.com> Karl, as i know there is no special timer that controls a SIP Session.I have the experience of media session open over 100 hours. One thing i can suggest u, Please check your session by session timing from ethereal capture. Check how much time each packet is taking for round trip, how much is jitter value etc etc. May be u can get a clue from that. Unless and until I see the packet capture I can't guess the reason.It is purely your environment and setting issue.Nothing related to SIP universal implementation. To get the details of timer used in SIP, U can go to RFC3261 and seach there. To make ur effort more easy search"Summary of timers". Thanks Badal Naik --- Karl Tian wrote: > Hello everyone, > Who can tell me if some rfc protocal(for > example: rfc3261) has define > the max duration of a sip conversation as 72 hours? > Now I'm testing the haleness for a kind of sip > client, but the > conversations of those clients all stop when the > duration reachs to 72 > hours. I guess that the question may be caused by a > special timer. > Please help me about this, thanks! > > > > > -- > Karl.Tian > Infinite Shanghai Communication Terminals Ltd. > Email :lingyunxjtu at gmail.com > Msn:lingyunxjtu at hotmail.com > > _______________________________________________ > This is the SIP Forum discussion mailing list > TO UNSUBSCRIBE, or edit your delivery options, > please visit > http://sipforum.org/mailman/listinfo/discussion > Post to the list at discussion at sipforum.org > ____________________________________________________________________________________ Shape Yahoo! in your own image. Join our Network Research Panel today! http://surveylink.yahoo.com/gmrs/yahoo_panel_invite.asp?a=7 From bn.darshan at gmail.com Thu Oct 4 08:26:23 2007 From: bn.darshan at gmail.com (darshan b n) Date: Thu, 4 Oct 2007 13:56:23 +0530 Subject: [SIPForum-discussion] sips message in ethereal In-Reply-To: <345812.11216.qm@web8320.mail.in.yahoo.com> References: <555d2a660710040054i231a803bua71e70caf2a731ae@mail.gmail.com> <345812.11216.qm@web8320.mail.in.yahoo.com> Message-ID: <555d2a660710040126l3c3bf225qab0b2e1872165607@mail.gmail.com> why it is not able to capture yahoo messages which runs on sip ? On 04/10/2007, Lakshminarayanan.Ramasami wrote: > > type in filter textbox like this sip > > *darshan b n * wrote: > > > Hi all, > > I want know how to capture a sips message in ethereal? > > If not is there any other free network protocol analyzer which support > this feature? > > Reply ASAP > > -- > Darshan B N > > Thanks & Regards > _______________________________________________ > This is the SIP Forum discussion mailing list > TO UNSUBSCRIBE, or edit your delivery options, please visit > http://sipforum.org/mailman/listinfo/discussion > Post to the list at discussion at sipforum.org > > > > > Thanks and Regards, > Lakshminarayanan.Ramasami > Cell : +91-9980840622,+91-80-41227491 > (BANGALORE) Cell : +91-9840264214,+91-4153-252636 (THIRUKKOYILUR) > > > ------------------------------ > Bring your gang together - do your thing. Start your group. > > -- Darshan B N Thanks & Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: From kunusan at yahoo.com Thu Oct 4 08:54:46 2007 From: kunusan at yahoo.com (badal naik) Date: Thu, 4 Oct 2007 01:54:46 -0700 (PDT) Subject: [SIPForum-discussion] sips message in ethereal In-Reply-To: <555d2a660710040054i231a803bua71e70caf2a731ae@mail.gmail.com> Message-ID: <864190.51307.qm@web56711.mail.re3.yahoo.com> If you have ethereal Installed in your system, u can follow below instructions, if dont know how to capture the packet. 1.Open The ethereal. 2.Click capture from the main menu. 3.Click Options. 4.In Options, Set the interface upon which you want to receive the packets.Basically it will be an ethernet card. 5. Then Initiate ur SIP activity.And see the contents in Ethereal Tool. 6. In ethereal, there will be one filter.Type there SIP and click Apply. Save this as .cab file if u want to save the file. For more details u can follow ethereal guidelines available at their URL. Others: Analyzer for win32. Sniffer(Sniffer technology) Etherpeek(From Wildpackets) Have not used any of them except ethereal. May be if u wish can do an R&D. Thanks Badal Naik --- darshan b n wrote: > Hi all, > > I want know how to capture a sips message in > ethereal? > > If not is there any other free network protocol > analyzer which support this > feature? > > Reply ASAP > > -- > Darshan B N > > Thanks & Regards > > _______________________________________________ > This is the SIP Forum discussion mailing list > TO UNSUBSCRIBE, or edit your delivery options, > please visit > http://sipforum.org/mailman/listinfo/discussion > Post to the list at discussion at sipforum.org > ____________________________________________________________________________________ Yahoo! oneSearch: Finally, mobile search that gives answers, not web links. http://mobile.yahoo.com/mobileweb/onesearch?refer=1ONXIC From victor.pascual.avila at gmail.com Thu Oct 4 09:09:56 2007 From: victor.pascual.avila at gmail.com (=?ISO-8859-1?Q?Victor_Pascual_=C1vila?=) Date: Thu, 4 Oct 2007 11:09:56 +0200 Subject: [SIPForum-discussion] sips message in ethereal In-Reply-To: <555d2a660710040126l3c3bf225qab0b2e1872165607@mail.gmail.com> References: <555d2a660710040054i231a803bua71e70caf2a731ae@mail.gmail.com> <345812.11216.qm@web8320.mail.in.yahoo.com> <555d2a660710040126l3c3bf225qab0b2e1872165607@mail.gmail.com> Message-ID: <618e24240710040209q6d3d487eh3d6b934272c27971@mail.gmail.com> Hi, use the filter 'sip' or capture by used port. Regards, Victor Pascual -------------- next part -------------- An HTML attachment was scrubbed... URL: From shakthi_msc at yahoo.com Thu Oct 4 09:20:16 2007 From: shakthi_msc at yahoo.com (shakthi_msc) Date: 4 Oct 2007 02:20:16 -0700 Subject: [SIPForum-discussion] Do we like the same books? Message-ID: <200710040920.l949KI4L011827@sipforum.org> I just joined Shelfari to connect with other book lovers. Come see the books I love and see if we have any in common. Then pick my next book so I can keep on reading. Click below to join my group of friends on Shelfari! http://www.shelfari.com/Register.aspx?ActivityId=22465602&InvitationCode=b46e3bcf-4169-42b2-bb6d-a9515f7ea3b5 shakthi_msc Shelfari is a free site that lets you share book ratings and reviews with friends and meet people who have similar tastes in books. It also lets you build an online bookshelf, join book clubs, and get good book recommendations from friends. You should check it out. -------- You have received this email because shakthi_msc (shakthi_msc at yahoo.com) directly invited you to join his/her community on Shelfari. It is against Shelfari's policies to invite people who you don't know directly. Follow this link (http://www.shelfari.com/actions/emailoptout.aspx?email=discussion at sipforum.org&activityid=22465602) to prevent future invitations to this address. If you believe you do not know this person, you may view (http://www.shelfari.com/shakthi_msc) his/her Shelfari page or report him/her in our feedback (http://www.shelfari.com/Feedback.aspx) section. Shelfari, 616 1st Ave #300, Seattle, WA 98104 -------------- next part -------------- An HTML attachment was scrubbed... URL: From radhashyambehera at gmail.com Thu Oct 4 10:38:35 2007 From: radhashyambehera at gmail.com (Radhashyam Behera) Date: Thu, 4 Oct 2007 16:08:35 +0530 Subject: [SIPForum-discussion] Requirement in RTP & IMS Team Message-ID: <7c4722c20710040338o2b385d87xc4d1f474f789d558@mail.gmail.com> Hi This is Radhashyam Behera, working with NetHawk Network INdia Pvt Ltd, India. We have urgent requirement in RTP and IMS Team. Interested candidate can apply with the same subject line to me. Detail requirement is given below: *RTP Team:* * * *Resource Requirement: 2* The candidate must have the below criteria. 1. 3+ years of experience. 2. Good C/C++ programming skill (application layer programming experience). 3. Socket programming, 4. Multithreaded programming experience. 5. Familiar with Linux environment. 6. Good knowledge of Real Time protocol, TCP/IP stacks. 7. Good Kernel programming skill and have experience in designing (module level designing experience will be enough) networking protocol state machines. * * *IMS Team:* * * *Resource Requirement: 2 * - Experience ? at least 2+yrs - Basic concepts of Protocols like Diameter, MEGACO, DNS, DHCP, XCAP, SOAP, HTTP, CAMEL, MGCP. - SIP protocol knowledge. - Good Knowledge in IMS Architecture with Interface idea. - Good communication skill - Good Documentation & Presentation. If the candidate knows the below tools, it will be added advantage: - Strong EAST Tool knowledge - Quality Center. - M5 and Ethereal Analyzer tool knowledge. With Regards, Radhashyam -------------- next part -------------- An HTML attachment was scrubbed... URL: From joel.silva at novabase.pt Thu Oct 4 11:31:56 2007 From: joel.silva at novabase.pt (Joel Silva) Date: Thu, 4 Oct 2007 12:31:56 +0100 Subject: [SIPForum-discussion] Registar a client Java References: <555d2a660710040054i231a803bua71e70caf2a731ae@mail.gmail.com><345812.11216.qm@web8320.mail.in.yahoo.com><555d2a660710040126l3c3bf225qab0b2e1872165607@mail.gmail.com> <618e24240710040209q6d3d487eh3d6b934272c27971@mail.gmail.com> Message-ID: Does anybody have a simple example in Java of a client registering in the registrar server? I?m only interested in the part of the client. Thanks, Joel. -------------- next part -------------- An HTML attachment was scrubbed... URL: From chahn at mytelepath.com Thu Oct 4 13:45:09 2007 From: chahn at mytelepath.com (Chris Hahn) Date: Thu, 04 Oct 2007 08:45:09 -0500 Subject: [SIPForum-discussion] Registar a client Java In-Reply-To: Message-ID: <731ec9f5e63a5844a889ca4565122ffc@mytelepath.com> Can I please be removed from this list? I've made several requests via the SIPForum website. Thanks, _____ From: discussion-bounces at sipforum.org [mailto:discussion-bounces at sipforum.org] On Behalf Of Joel Silva Sent: Thursday, October 04, 2007 6:32 AM To: discussion at sipforum.org Subject: [SIPForum-discussion] Registar a client Java Does anybody have a simple example in Java of a client registering in the registrar server? I?m only interested in the part of the client. Thanks, Joel. -------------- next part -------------- An HTML attachment was scrubbed... URL: From raymond.jender.ctr at disa.mil Thu Oct 4 14:37:27 2007 From: raymond.jender.ctr at disa.mil (Jender, Raymond C CTR DISA JITC) Date: Thu, 4 Oct 2007 07:37:27 -0700 Subject: [SIPForum-discussion] Asterisk Usage (UNCLASSIFIED) In-Reply-To: <40E89886C8B3B54B98C5291646C591AA01B6E54A@ZMY16EXM67.ds.mot.com> References: <40E89886C8B3B54B98C5291646C591AA01B6E54A@ZMY16EXM67.ds.mot.com> Message-ID: <00E9117C300386479B7E2F3CDBF798F601E7EC62@pothia.disanet.disa-u.mil> Classification: UNCLASSIFIED Caveats: NONE Go to http://www.asterisk.org Raymond C. Jender Booz|Allen|Hamilton DSN IA Test Team Ft. Huachuca, Az. 520-538-2588 -----Original Message----- From: discussion-bounces at sipforum.org [mailto:discussion-bounces at sipforum.org] On Behalf Of S Selvakumar-VPFR47 Sent: Wednesday, October 03, 2007 11:57 PM To: discussion at sipforum.org Subject: [SIPForum-discussion] Asterisk Usage Hi, I want to know some info about Asterisk. Can anyone help me regards Selva Classification: UNCLASSIFIED Caveats: NONE From abdel_mameri at hotmail.com Thu Oct 4 14:55:18 2007 From: abdel_mameri at hotmail.com (mameri abdelhamid) Date: Thu, 04 Oct 2007 14:55:18 +0000 Subject: [SIPForum-discussion] Can I please be removed from this list? In-Reply-To: <731ec9f5e63a5844a889ca4565122ffc@mytelepath.com> Message-ID: An HTML attachment was scrubbed... URL: From bn.darshan at gmail.com Fri Oct 5 07:47:05 2007 From: bn.darshan at gmail.com (darshan b n) Date: Fri, 5 Oct 2007 13:17:05 +0530 Subject: [SIPForum-discussion] Rtp testing tool Message-ID: <555d2a660710050047t308cd98dicbf4462c70a68901@mail.gmail.com> Hi all, I want to test RTP protocol (g.711,g.729..........AMR)..........Can you please suggest open souce tool which gives statistics such as MOS ,JITTER,LATENCY....etc -- Darshan B N Thanks & Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: From atutush at yahoo.com Fri Oct 5 08:55:16 2007 From: atutush at yahoo.com (ahmet tutus) Date: Fri, 5 Oct 2007 11:55:16 +0300 (EEST) Subject: [SIPForum-discussion] ACK problem Message-ID: <268147.96922.qm@web53303.mail.re2.yahoo.com> Hi all. I send ACK from Client1 to Client2, and Client 2 can not receive the ACK. When I look at the logs, I see that there are 4 ACKs in the system, but there is no tag in the To header part of the last ACK. I use Route:[routes] keyword and "rrs=true" in the last received OK part. I take also some INFO messages. Why do I take it? Any error in these codes? Any idea about this situation? Thanks in advance, Best Regards... Ahmet TUTUS TURKEY Bogazici University System Control Engineering --------------------------------- Yahoo! kullaniyor musunuz? Istenmeyen postadan biktiniz mi? Istenmeyen postadan en iyi korunma Yahoo! Posta'da http://tr.mail.yahoo.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From somi.suresh at gmail.com Fri Oct 5 09:41:33 2007 From: somi.suresh at gmail.com (cheetah) Date: Fri, 5 Oct 2007 15:11:33 +0530 Subject: [SIPForum-discussion] AMR codec required for the softphone Message-ID: <86f9a8d10710050241x7fea04v1e944c5fd4eba76d@mail.gmail.com> Hi, I need softphone which enabled with AMR Codec. If anybody has details reply to me ASAP From smanickam at velankani.com Fri Oct 5 10:07:40 2007 From: smanickam at velankani.com (Shankar Manickam) Date: Fri, 5 Oct 2007 15:37:40 +0530 Subject: [SIPForum-discussion] CSeq in re-Invite Message-ID: <010d01c80737$906edf00$0e1c000a@blr.velankani.com> Hi, Is there any situation where cseq will get incremented for re-Invite? With Regards, Shankar Manickam -------------- next part -------------- An HTML attachment was scrubbed... URL: From radhashyambehera at gmail.com Fri Oct 5 10:45:10 2007 From: radhashyambehera at gmail.com (Radhashyam Behera) Date: Fri, 5 Oct 2007 16:15:10 +0530 Subject: [SIPForum-discussion] CSeq in re-Invite In-Reply-To: <010d01c80737$906edf00$0e1c000a@blr.velankani.com> References: <010d01c80737$906edf00$0e1c000a@blr.velankani.com> Message-ID: <7c4722c20710050345h3feac90xdf09b381614ffcc2@mail.gmail.com> In case of re-Invite CSeq needs to be incremented always for the same Dialog. Thanks & Regards, Radhashyam On 10/5/07, Shankar Manickam wrote: > > Hi, > > Is there any situation where cseq will get incremented for re-Invite? > > > > > > *With Regards,* > > *Shankar Manickam* > > > > _______________________________________________ > This is the SIP Forum discussion mailing list > TO UNSUBSCRIBE, or edit your delivery options, please visit > http://sipforum.org/mailman/listinfo/discussion > Post to the list at discussion at sipforum.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bn.darshan at gmail.com Fri Oct 5 11:16:17 2007 From: bn.darshan at gmail.com (darshan b n) Date: Fri, 5 Oct 2007 16:46:17 +0530 Subject: [SIPForum-discussion] CSeq in re-Invite In-Reply-To: <7c4722c20710050345h3feac90xdf09b381614ffcc2@mail.gmail.com> References: <010d01c80737$906edf00$0e1c000a@blr.velankani.com> <7c4722c20710050345h3feac90xdf09b381614ffcc2@mail.gmail.com> Message-ID: <555d2a660710050416g2c9af425l50300597c32190f5@mail.gmail.com> Call hold feature where u can find re-invite with cseq : incremented Regards darshan b n On 05/10/2007, Radhashyam Behera wrote: > > In case of re-Invite CSeq needs to be incremented always for the same > Dialog. > > Thanks & Regards, > Radhashyam > > On 10/5/07, Shankar Manickam wrote: > > > Hi, > > > > Is there any situation where cseq will get incremented for > > re-Invite? > > > > > > > > > > > > *With Regards,* > > > > *Shankar Manickam* > > > > > > > > _______________________________________________ > > This is the SIP Forum discussion mailing list > > TO UNSUBSCRIBE, or edit your delivery options, please visit > > http://sipforum.org/mailman/listinfo/discussion > > Post to the list at discussion at sipforum.org > > > > > > _______________________________________________ > This is the SIP Forum discussion mailing list > TO UNSUBSCRIBE, or edit your delivery options, please visit > http://sipforum.org/mailman/listinfo/discussion > Post to the list at discussion at sipforum.org > > -- Darshan B N Thanks & Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: From shiv.kumr at gmail.com Fri Oct 5 11:51:25 2007 From: shiv.kumr at gmail.com (Shiv) Date: Fri, 5 Oct 2007 17:21:25 +0530 Subject: [SIPForum-discussion] CSeq in re-Invite In-Reply-To: <555d2a660710050416g2c9af425l50300597c32190f5@mail.gmail.com> References: <010d01c80737$906edf00$0e1c000a@blr.velankani.com> <7c4722c20710050345h3feac90xdf09b381614ffcc2@mail.gmail.com> <555d2a660710050416g2c9af425l50300597c32190f5@mail.gmail.com> Message-ID: <2a6a20ec0710050451q2ab42047o9e368a83b4bfe841@mail.gmail.com> I suppose, three re-invite will be involved for the same dialog on scenario like : a. hold b. conecting to Music c. connecting back to the endpoint, Since it is same dialog, Cseq will be incremented. On 10/5/07, darshan b n wrote: > > Call hold feature where u can find re-invite with cseq : incremented > > Regards > darshan b n > > > On 05/10/2007, Radhashyam Behera wrote: > > > > In case of re-Invite CSeq needs to be incremented always for the same > > Dialog. > > > > Thanks & Regards, > > Radhashyam > > > > On 10/5/07, Shankar Manickam < smanickam at velankani.com> wrote: > > > > > Hi, > > > > > > Is there any situation where cseq will get incremented for > > > re-Invite? > > > > > > > > > > > > > > > > > > *With Regards,* > > > > > > *Shankar Manickam* > > > > > > > > > > > > _______________________________________________ > > > This is the SIP Forum discussion mailing list > > > TO UNSUBSCRIBE, or edit your delivery options, please visit > > > http://sipforum.org/mailman/listinfo/discussion > > > Post to the list at discussion at sipforum.org > > > > > > > > > > _______________________________________________ > > This is the SIP Forum discussion mailing list > > TO UNSUBSCRIBE, or edit your delivery options, please visit > > http://sipforum.org/mailman/listinfo/discussion > > Post to the list at discussion at sipforum.org > > > > > > > -- > Darshan B N > > Thanks & Regards > _______________________________________________ > This is the SIP Forum discussion mailing list > TO UNSUBSCRIBE, or edit your delivery options, please visit > http://sipforum.org/mailman/listinfo/discussion > Post to the list at discussion at sipforum.org > > -- With best regards, Sivakumar Arumugam +91-9282419188 Alcatel-Lucent. -------------- next part -------------- An HTML attachment was scrubbed... URL: From VPFR47 at motorola.com Fri Oct 5 15:33:26 2007 From: VPFR47 at motorola.com (S Selvakumar-VPFR47) Date: Fri, 5 Oct 2007 23:33:26 +0800 Subject: [SIPForum-discussion] CSeq in re-Invite In-Reply-To: <010d01c80737$906edf00$0e1c000a@blr.velankani.com> Message-ID: <40E89886C8B3B54B98C5291646C591AA01BA21FA@ZMY16EXM67.ds.mot.com> Hi Shankar, Try a Call Hold scenario, where you can find Cseq gets incremented for Hold Invite and UnHold Invite regards Selva ________________________________ From: discussion-bounces at sipforum.org [mailto:discussion-bounces at sipforum.org] On Behalf Of Shankar Manickam Sent: Friday, October 05, 2007 3:38 PM To: discussion at sipforum.org Subject: [SIPForum-discussion] CSeq in re-Invite Hi, Is there any situation where cseq will get incremented for re-Invite? With Regards, Shankar Manickam -------------- next part -------------- An HTML attachment was scrubbed... URL: From losmacocos_1 at hotmail.com Sat Oct 6 08:24:43 2007 From: losmacocos_1 at hotmail.com (Ignacio Macocos) Date: Sat, 6 Oct 2007 08:24:43 +0000 Subject: [SIPForum-discussion] Can I please be removed from this list? In-Reply-To: References: <731ec9f5e63a5844a889ca4565122ffc@mytelepath.com> Message-ID: Please me too!!!!! I'dont like to receive mails from this forum. thks From: abdel_mameri at hotmail.comTo: chahn at mytelepath.com; discussion at sipforum.orgDate: Thu, 4 Oct 2007 14:55:18 +0000Subject: [SIPForum-discussion] Can I please be removed from this list? Can I please be removed from this list? I?ve made several requests via the SIPForum website. Thanks, From: "Chris Hahn" To: "discussion at sipforum.org" Subject: Re: [SIPForum-discussion] Registar a client JavaDate: Thu, 04 Oct 2007 08:45:09 -0500 Can I please be removed from this list? I?ve made several requests via the SIPForum website. Thanks, From: discussion-bounces at sipforum.org [mailto:discussion-bounces at sipforum.org] On Behalf Of Joel SilvaSent: Thursday, October 04, 2007 6:32 AMTo: discussion at sipforum.orgSubject: [SIPForum-discussion] Registar a client Java Does anybody have a simple example in Java of a client registering in the registrar server? I?m only interested in the part of the client. Thanks, Joel. >_______________________________________________>This is the SIP Forum discussion mailing list>TO UNSUBSCRIBE, or edit your delivery options, please visit http://sipforum.org/mailman/listinfo/discussion>Post to the list at discussion at sipforum.org Saviez-vous que Windows Live Messenger est disponible d?s maintenant sur votre GSM ? _________________________________________________________________ Climb to the top of the charts!? Play Star Shuffle:? the word scramble challenge with star power. http://club.live.com/star_shuffle.aspx?icid=starshuffle_wlmailtextlink_oct -------------- next part -------------- An HTML attachment was scrubbed... URL: From amit.v at pyronetworks.com Sat Oct 6 13:08:06 2007 From: amit.v at pyronetworks.com (amit) Date: Sat, 06 Oct 2007 18:38:06 +0530 Subject: [SIPForum-discussion] images with sip messages Message-ID: <1191676086.6009.18.camel@amit> Hi All, Can we send images with sip messages ? if yes, then how it is possible ? Thanks in Advance Amit -------------- next part -------------- An HTML attachment was scrubbed... URL: From amit.v at pyronetworks.com Sat Oct 6 13:10:56 2007 From: amit.v at pyronetworks.com (amit) Date: Sat, 06 Oct 2007 18:40:56 +0530 Subject: [SIPForum-discussion] confrence in sip Message-ID: <1191676256.6009.21.camel@amit> Hi All, I am working on Sip Chat Sever. Sip can support chat conference?????? Thanks in Advance Amit From eshwarry at gmail.com Sun Oct 7 03:02:00 2007 From: eshwarry at gmail.com (eswari s) Date: Sun, 7 Oct 2007 08:32:00 +0530 Subject: [SIPForum-discussion] Which RFC ? Message-ID: <83ea91a60710062002q6f0a9ceev91587bb8a693404b@mail.gmail.com> Hi Which RFC is followed for file transfer in sip phone's ?? best wishes, eshwary -------------- next part -------------- An HTML attachment was scrubbed... URL: From triveni.prabhu at wipro.com Sun Oct 7 04:49:59 2007 From: triveni.prabhu at wipro.com (triveni.prabhu at wipro.com) Date: Sun, 7 Oct 2007 10:19:59 +0530 Subject: [SIPForum-discussion] Which RFC ? References: <83ea91a60710062002q6f0a9ceev91587bb8a693404b@mail.gmail.com> Message-ID: <4EB051147731D64B9B9187139BAC3E9502ACDE@BLR-SJP-MBX01.wipro.com> Hi, You can refer to http://tools.ietf.org/id/draft-isomaki-sipping-file-transfer-00.txt Thank you, Regards, Triveni Prabhu. ________________________________ From: discussion-bounces at sipforum.org on behalf of eswari s Sent: Sun 10/7/2007 8:32 AM To: discussion at sipforum.org Subject: [SIPForum-discussion] Which RFC ? Hi Which RFC is followed for file transfer in sip phone's ?? best wishes, eshwary -------------- next part -------------- An HTML attachment was scrubbed... URL: From harini.dhanasekaran at wipro.com Sun Oct 7 05:49:56 2007 From: harini.dhanasekaran at wipro.com (harini.dhanasekaran at wipro.com) Date: Sun, 7 Oct 2007 11:19:56 +0530 Subject: [SIPForum-discussion] confrence in sip References: <1191676256.6009.21.camel@amit> Message-ID: Hi Amit, Yes, SIP can support chat or IM conference. Refer to this link: http://www.ietf.org/internet-drafts/draft-ietf-simple-chat-00.txt Note: The link above is an internet draft document valid till dec 14, 2007. Thank you, Regards, Harini Dhanasekaran. ________________________________ From: discussion-bounces at sipforum.org on behalf of amit Sent: Sat 10/6/2007 6:40 PM To: discussion at sipforum.org Subject: [SIPForum-discussion] confrence in sip Hi All, I am working on Sip Chat Sever. Sip can support chat conference?????? Thanks in Advance Amit _______________________________________________ This is the SIP Forum discussion mailing list TO UNSUBSCRIBE, or edit your delivery options, please visit http://sipforum.org/mailman/listinfo/discussion Post to the list at discussion at sipforum.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From sivam at motorola.com Sun Oct 7 08:26:45 2007 From: sivam at motorola.com (Siva M-Q16748) Date: Sun, 7 Oct 2007 16:26:45 +0800 Subject: [SIPForum-discussion] Out of Office AutoReply: Do we like the same books? Message-ID: <988EE2C769AC284ABAE9328BFC10703F296053@ZMY16EXM66.ds.mot.com> Hi , I am Out Of Office till 09-Oct-2007 . I will be able to reply to your mail only on 10-Oct-2007 . During this time please contact my manager Rao K Venkateswara - Q16395 for any issues. Best Regards, Siva M Mobile : 9880108336 -------------- next part -------------- An HTML attachment was scrubbed... URL: From baolovebao at gmail.com Sun Oct 7 11:38:50 2007 From: baolovebao at gmail.com (Donald Lee) Date: Sun, 7 Oct 2007 19:38:50 +0800 Subject: [SIPForum-discussion] Rtp testing tool In-Reply-To: <555d2a660710050047t308cd98dicbf4462c70a68901@mail.gmail.com> References: <555d2a660710050047t308cd98dicbf4462c70a68901@mail.gmail.com> Message-ID: I think ethereal tool can meet part of your requirement. On 10/5/07, darshan b n wrote: > > Hi all, > > I want to test RTP protocol (g.711,g.729..........AMR)..........Can you > please suggest open souce tool which gives statistics such as MOS > ,JITTER,LATENCY....etc > > > -- > Darshan B N > > Thanks & Regards > > _______________________________________________ > This is the SIP Forum discussion mailing list > TO UNSUBSCRIBE, or edit your delivery options, please visit > http://sipforum.org/mailman/listinfo/discussion > Post to the list at discussion at sipforum.org > > -- BR Donald -------------- next part -------------- An HTML attachment was scrubbed... URL: From lingyunxjtu at gmail.com Mon Oct 8 00:40:31 2007 From: lingyunxjtu at gmail.com (Karl Tian) Date: Mon, 8 Oct 2007 08:40:31 +0800 Subject: [SIPForum-discussion] The max duration of SIP conversation In-Reply-To: <200842.58269.qm@web56703.mail.re3.yahoo.com> References: <807efb400710032342rdb8ca9fyc456c7a3ab622060@mail.gmail.com> <200842.58269.qm@web56703.mail.re3.yahoo.com> Message-ID: <807efb400710071740r20fb63aej93c5119380c77d36@mail.gmail.com> Hi badal, Thanks for your reply to my question. Now I've oriented the reason for sip conversation disconnect once duration reaching to 72 hours. The SIP server I used for testing is of ONDO types, and there is a configuration "Talking timeout" with default value "259200000ms", and I will make sure that by capturing packets. Best wishes Your Karl On 10/4/07, badal naik wrote: > > Karl, > as i know there is no special timer that controls a > SIP Session.I have the experience of media session > open over 100 hours. > One thing i can suggest u, Please check your session > by session timing from ethereal capture. > Check how much time each packet is taking for round > trip, how much is jitter value etc etc. > > May be u can get a clue from that. Unless and until I > see the packet capture I can't guess the reason.It is > purely your environment and setting issue.Nothing > related to SIP universal implementation. > > To get the details of timer used in SIP, U can go to > RFC3261 and seach there. > To make ur effort more easy search"Summary of timers". > > Thanks > Badal Naik > --- Karl Tian wrote: > > > Hello everyone, > > Who can tell me if some rfc protocal(for > > example: rfc3261) has define > > the max duration of a sip conversation as 72 hours? > > Now I'm testing the haleness for a kind of sip > > client, but the > > conversations of those clients all stop when the > > duration reachs to 72 > > hours. I guess that the question may be caused by a > > special timer. > > Please help me about this, thanks! > > > > > > > > > > -- > > Karl.Tian > > Infinite Shanghai Communication Terminals Ltd. > > Email :lingyunxjtu at gmail.com > > Msn:lingyunxjtu at hotmail.com > > > _______________________________________________ > > This is the SIP Forum discussion mailing list > > TO UNSUBSCRIBE, or edit your delivery options, > > please visit > > http://sipforum.org/mailman/listinfo/discussion > > Post to the list at discussion at sipforum.org > > > > > > > ____________________________________________________________________________________ > Shape Yahoo! in your own image. Join our Network Research Panel today! > http://surveylink.yahoo.com/gmrs/yahoo_panel_invite.asp?a=7 > > > -- Karl.Tian Infinite Shanghai Communication Terminals Ltd. Mobile:(+86)15902148975 Office :(+86)21-38954999-775 Email :lingyunxjtu at gmail.com Msn:lingyunxjtu at hotmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From Richard.Agonias at digitel.ph Mon Oct 8 01:49:41 2007 From: Richard.Agonias at digitel.ph (Agonias, Richard L. (Digitel-GSM)) Date: Mon, 8 Oct 2007 09:49:41 +0800 Subject: [SIPForum-discussion] AMR In-Reply-To: <807efb400710071740r20fb63aej93c5119380c77d36@mail.gmail.com> Message-ID: <90B7FB18EBCD424B83BA9FEA0D4319E8337664@dgtlmail.digitel.ph> Hi All, Good day! I know you guys used AMR already. Does anyone here know if the 39 bits on the AMR SID occurs every 20ms or 160ms? RFC 3267 did not specify it. Thanks! -richard -------------- next part -------------- An HTML attachment was scrubbed... URL: From govindraj_h at yahoo.co.in Mon Oct 8 03:31:58 2007 From: govindraj_h at yahoo.co.in (Govindraj.B.H @ Gkk) Date: Mon, 8 Oct 2007 09:01:58 +0530 (IST) Subject: [SIPForum-discussion] Asterisk Usage Message-ID: <426133.56377.qm@web8414.mail.in.yahoo.com> Hi Selva, I think there are 2 types of Asterisks One is " * " that means ALL and another is " $ " that means ANY. If any body knows more about this help us. Thanks and Regards Govindraj B H ----- Original Message ---- From: S Selvakumar-VPFR47 To: discussion at sipforum.org Sent: Thursday, 4 October, 2007 2:56:32 AM Subject: [SIPForum-discussion] Asterisk Usage Hi, I want to know some info about Asterisk. Can anyone help me regards Selva Explore your hobbies and interests. Go to http://in.promos.yahoo.com/groups -------------- next part -------------- An HTML attachment was scrubbed... URL: From devanand at TechMahindra.com Mon Oct 8 04:51:31 2007 From: devanand at TechMahindra.com (Devanand Kumar) Date: Mon, 8 Oct 2007 10:21:31 +0530 Subject: [SIPForum-discussion] Asterisk Usage Message-ID: <089781E831473740B23334AE52636CD30825A60B@SINBNGEX001.TechMahindra.com> Hi, Attached document will give a brief introduction of Asterisk PBX software. If U want more information u can follow the following link. http://www.digium.com . http://www.asterisk.org/ Thanks and Regards, Devanand Kumar ________________________________ From: discussion-bounces at sipforum.org [mailto:discussion-bounces at sipforum.org] On Behalf Of Govindraj.B.H @ Gkk Sent: Monday, October 08, 2007 9:02 AM To: S Selvakumar-VPFR47; discussion at sipforum.org Subject: Re: [SIPForum-discussion] Asterisk Usage Hi Selva, I think there are 2 types of Asterisks One is " * " that means ALL and another is " $ " that means ANY. If any body knows more about this help us. Thanks and Regards Govindraj B H ----- Original Message ---- From: S Selvakumar-VPFR47 To: discussion at sipforum.org Sent: Thursday, 4 October, 2007 2:56:32 AM Subject: [SIPForum-discussion] Asterisk Usage Hi, I want to know some info about Asterisk. Can anyone help me regards Selva ________________________________ Chat on a cool, new interface. No download required. Click here. ============================================================================================================================ Disclaimer: This message and the information contained herein is proprietary and confidential and subject to the Tech Mahindra policy statement, you may review the policy at http://www.techmahindra.com/Disclaimer.html externally and http://tim.techmahindra.com/Disclaimer.html internally within Tech Mahindra. ============================================================================================================================ -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: asterisk.rtf Type: application/rtf Size: 5035 bytes Desc: asterisk.rtf URL: From wellya at wellya.net Mon Oct 8 05:09:05 2007 From: wellya at wellya.net (Stewart.Zhong) Date: Mon, 8 Oct 2007 13:09:05 +0800 (CST) Subject: [SIPForum-discussion] who can send me about QOS training file? Message-ID: <22616909.1191820145327.JavaMail.postfix@l_010_011_015_034> who can send me about QOS training file? or QOS related specs? Very deelply Thanks!!! -------------- next part -------------- An HTML attachment was scrubbed... URL: From mishra.abhishek at wipro.com Mon Oct 8 06:30:46 2007 From: mishra.abhishek at wipro.com (mishra.abhishek at wipro.com) Date: Mon, 8 Oct 2007 12:00:46 +0530 Subject: [SIPForum-discussion] images with sip messages In-Reply-To: <1191676086.6009.18.camel@amit> References: <1191676086.6009.18.camel@amit> Message-ID: <01949ABCDD38EF4A95B8C347C399D1CE057C5FAB@BLR-EC-MBX02.wipro.com> Through the MESSAGE message you can send any attachment. Just check the RFC2778/2779. ________________________________ From: discussion-bounces at sipforum.org [mailto:discussion-bounces at sipforum.org] On Behalf Of amit Sent: Saturday, October 06, 2007 6:38 PM To: discussion at sipforum.org Subject: [SIPForum-discussion] images with sip messages Hi All, Can we send images with sip messages ? if yes, then how it is possible ? Thanks in Advance Amit -------------- next part -------------- An HTML attachment was scrubbed... URL: From chauhan_delhi at yahoo.com Mon Oct 8 11:03:59 2007 From: chauhan_delhi at yahoo.com (Ramesh Chauhan) Date: Mon, 8 Oct 2007 04:03:59 -0700 (PDT) Subject: [SIPForum-discussion] Asterisk- cisco Call Menager Message-ID: <843093.78935.qm@web34411.mail.mud.yahoo.com> Hi, I am getting "503-Service Not Available" from my CCM(Ciso Call Manager). Ethreal logs of asterisk is attached. We have extensions on my CCM and calls are going and comming perfectly fine to CCM extensions. We are ths error message ony when wants to dial ISD number through CCM. From CCM we are able to make ISD Calls. Can anybody help me . why i am getting this error message. For any other information, please free to revert back. Thanking you in anticipation. Regards Chauhan with regards Ramesh Chauhan --------------------------------- Yahoo! oneSearch: Finally, mobile search that gives answers, not web links. -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: sip_call_503.zip Type: application/x-zip-compressed Size: 2058 bytes Desc: 2297908990-sip_call_503.zip URL: From victor.pascual.avila at gmail.com Mon Oct 8 12:10:32 2007 From: victor.pascual.avila at gmail.com (=?ISO-8859-1?Q?Victor_Pascual_=C1vila?=) Date: Mon, 8 Oct 2007 14:10:32 +0200 Subject: [SIPForum-discussion] who can send me about QOS training file? In-Reply-To: <22616909.1191820145327.JavaMail.postfix@l_010_011_015_034> References: <22616909.1191820145327.JavaMail.postfix@l_010_011_015_034> Message-ID: <618e24240710080510x716997b0yc817ea574072724f@mail.gmail.com> Here you are http://www.ssuet.edu.pk/~amkhan/cisco/Cisco_IP_Telephony_QoS_Design_Guide.pdf Victor Pascual On 08/10/2007, Stewart.Zhong wrote: > > > who can send me about QOS training file? or QOS related specs? > > Very deelply Thanks!!! > _______________________________________________ > This is the SIP Forum discussion mailing list > TO UNSUBSCRIBE, or edit your delivery options, please visit > http://sipforum.org/mailman/listinfo/discussion > Post to the list at discussion at sipforum.org > > From smanickam at velankani.com Mon Oct 8 12:53:39 2007 From: smanickam at velankani.com (Shankar Manickam) Date: Mon, 8 Oct 2007 18:23:39 +0530 Subject: [SIPForum-discussion] CSeq in re-Invite In-Reply-To: <40E89886C8B3B54B98C5291646C591AA01BA21FA@ZMY16EXM67.ds.mot.com> Message-ID: <00d401c809aa$3fb86ac0$0e1c000a@blr.velankani.com> Hi All, I am not seeing that cseq gets incremented for Call hold in the given link http://tech-invite.com/Ti-sip-service-1.html Is there any special condition to get it? With regards, Shankar. -----Original Message----- From: S Selvakumar-VPFR47 [mailto:VPFR47 at motorola.com] Sent: Friday, October 05, 2007 9:03 PM To: Shankar Manickam; discussion at sipforum.org Subject: RE: [SIPForum-discussion] CSeq in re-Invite Hi Shankar, Try a Call Hold scenario, where you can find Cseq gets incremented for Hold Invite and UnHold Invite regards Selva _____ From: discussion-bounces at sipforum.org [mailto:discussion-bounces at sipforum.org] On Behalf Of Shankar Manickam Sent: Friday, October 05, 2007 3:38 PM To: discussion at sipforum.org Subject: [SIPForum-discussion] CSeq in re-Invite Hi, Is there any situation where cseq will get incremented for re-Invite? With Regards, Shankar Manickam -------------- next part -------------- An HTML attachment was scrubbed... URL: From sumantasen at tataelxsi.co.in Tue Oct 9 09:59:10 2007 From: sumantasen at tataelxsi.co.in (Sumanta Sen) Date: Tue, 9 Oct 2007 15:29:10 +0530 Subject: [SIPForum-discussion] SIGCOMP Message-ID: <003501c80a5b$0a5ce160$f829320a@telxsi.com> Hi All, The TS standards suggest that SDP body should not be compressed. So it means that SIP headers are compressed and message body is not. How to decompress such messages ? How should the receiver distinguish between compressed and uncompressed parts of message? Sumanta The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments contained in it. Contact your Administrator for further information. From anurgupta at yahoo.com Tue Oct 9 13:31:52 2007 From: anurgupta at yahoo.com (anurag gupta) Date: Tue, 9 Oct 2007 06:31:52 -0700 (PDT) Subject: [SIPForum-discussion] CSeq in re-Invite In-Reply-To: <00d401c809aa$3fb86ac0$0e1c000a@blr.velankani.com> Message-ID: <34402.17906.qm@web63706.mail.re1.yahoo.com> Hi The RE-INVITE is generated from the callee side, so it can send any CSeq in its generated INVITE or any other message. But if the Re-INVITe is generated from the caller's end, then CSeq should be incremented. Regards Anurag Shankar Manickam wrote: Hi All, I am not seeing that cseq gets incremented for Call hold in the given link http://tech-invite.com/Ti-sip-service-1.html Is there any special condition to get it? With regards, Shankar. -----Original Message----- From: S Selvakumar-VPFR47 [mailto:VPFR47 at motorola.com] Sent: Friday, October 05, 2007 9:03 PM To: Shankar Manickam; discussion at sipforum.org Subject: RE: [SIPForum-discussion] CSeq in re-Invite Hi Shankar, Try a Call Hold scenario, where you can find Cseq gets incremented for Hold Invite and UnHold Invite regards Selva --------------------------------- From: discussion-bounces at sipforum.org [mailto:discussion-bounces at sipforum.org] On Behalf Of Shankar Manickam Sent: Friday, October 05, 2007 3:38 PM To: discussion at sipforum.org Subject: [SIPForum-discussion] CSeq in re-Invite Hi, Is there any situation where cseq will get incremented for re-Invite? With Regards, Shankar Manickam _______________________________________________ This is the SIP Forum discussion mailing list TO UNSUBSCRIBE, or edit your delivery options, please visit http://sipforum.org/mailman/listinfo/discussion Post to the list at discussion at sipforum.org --------------------------------- Be a better Globetrotter. Get better travel answers from someone who knows. Yahoo! Answers - Check it out. -------------- next part -------------- An HTML attachment was scrubbed... URL: From Adrian.Felea at mtsallstream.com Tue Oct 9 15:24:28 2007 From: Adrian.Felea at mtsallstream.com (Felea, Adrian) Date: Tue, 9 Oct 2007 11:24:28 -0400 Subject: [SIPForum-discussion] Caller identity privacy Message-ID: <2059B5D3BB19464A80466224E5F1B55901C59C9C@TJ1EXB02.mtsallstream.com> I am having issue with When dialing *67 from a PSTN phone (to hide the identity) with a name associated, this gets translated to an "ANONYMOUS" name within the "From" sip header. In this case the identity of the caller is not presented to the called party. When dialing the same from a PSTN phone with no name associated, the "ANONYMOUS" name is not part of the "From" sip header anymore and the identity of the caller is presented to the called party. In the RFC 3261 it states: The From header field allows for a display name. A UAC SHOULD use the display name "Anonymous", along with a syntactically correct, but otherwise meaningless URI (like sip:thisis at anonymous.invalid), if the identity of the client is to remain hidden. If I understand this correctly, does this mean that when calling from a PSTN phone with no name associated (and dialing *67 for privacy), the identity of the caller is still presented to the called party because we do not have an "ANONYMOUS" name within the "From" sip header? Thanks, Adrian -------------- next part -------------- An HTML attachment was scrubbed... URL: From nazeema_guttur at yahoo.com Wed Oct 10 11:06:09 2007 From: nazeema_guttur at yahoo.com (nazeema Tasneem) Date: Wed, 10 Oct 2007 04:06:09 -0700 (PDT) Subject: [SIPForum-discussion] security testing Message-ID: <147034.3651.qm@web37506.mail.mud.yahoo.com> Hi all, can anyone tell me about automated tools for testing security considerations(message flooding, registration hijak, call teardown etc) in SIP. Thanks Nazeema ____________________________________________________________________________________ Building a website is a piece of cake. Yahoo! Small Business gives you all the tools to get online. http://smallbusiness.yahoo.com/webhosting From eric.fiorini at u4eatech.com Wed Oct 10 11:39:09 2007 From: eric.fiorini at u4eatech.com (Eric Fiorini) Date: Wed, 10 Oct 2007 13:39:09 +0200 Subject: [SIPForum-discussion] security testing In-Reply-To: <147034.3651.qm@web37506.mail.mud.yahoo.com> Message-ID: <001401c80b32$2eafe990$ed001cac@u4eatech.com> Try this link ... I don't know if this is what you are looking for ... http://www.cert.org/advisories/CA-2003-06.html Eric Fiorini U4EA Technologies -----Original Message----- From: discussion-bounces at sipforum.org [mailto:discussion-bounces at sipforum.org] On Behalf Of nazeema Tasneem Sent: mercredi 10 octobre 2007 13:06 To: discussion at sipforum.org Subject: [SIPForum-discussion] security testing Hi all, can anyone tell me about automated tools for testing security considerations(message flooding, registration hijak, call teardown etc) in SIP. Thanks Nazeema ____________________________________________________________________________ ________ Building a website is a piece of cake. Yahoo! Small Business gives you all the tools to get online. http://smallbusiness.yahoo.com/webhosting _______________________________________________ This is the SIP Forum discussion mailing list TO UNSUBSCRIBE, or edit your delivery options, please visit http://sipforum.org/mailman/listinfo/discussion Post to the list at discussion at sipforum.org -- U4EA Technologies http://www.u4eatech.com From raymond.jender.ctr at disa.mil Wed Oct 10 14:47:41 2007 From: raymond.jender.ctr at disa.mil (Jender, Raymond C CTR DISA JITC) Date: Wed, 10 Oct 2007 07:47:41 -0700 Subject: [SIPForum-discussion] security testing (UNCLASSIFIED) In-Reply-To: <147034.3651.qm@web37506.mail.mud.yahoo.com> References: <147034.3651.qm@web37506.mail.mud.yahoo.com> Message-ID: <00E9117C300386479B7E2F3CDBF798F601E7EC77@pothia.disanet.disa-u.mil> Classification: UNCLASSIFIED Caveats: NONE Here is a comprehensive list of tools available. Have fun.... http://www.voipsa.org/Resources/tools.php Ray -----Original Message----- From: discussion-bounces at sipforum.org [mailto:discussion-bounces at sipforum.org] On Behalf Of nazeema Tasneem Sent: Wednesday, October 10, 2007 4:06 AM To: discussion at sipforum.org Subject: [SIPForum-discussion] security testing Hi all, can anyone tell me about automated tools for testing security considerations(message flooding, registration hijak, call teardown etc) in SIP. Thanks Nazeema ________________________________________________________________________ ____________ Building a website is a piece of cake. Yahoo! Small Business gives you all the tools to get online. http://smallbusiness.yahoo.com/webhosting _______________________________________________ This is the SIP Forum discussion mailing list TO UNSUBSCRIBE, or edit your delivery options, please visit http://sipforum.org/mailman/listinfo/discussion Post to the list at discussion at sipforum.org Classification: UNCLASSIFIED Caveats: NONE From tmaufer at musecurity.com Wed Oct 10 15:46:55 2007 From: tmaufer at musecurity.com (Thomas Maufer) Date: Wed, 10 Oct 2007 08:46:55 -0700 Subject: [SIPForum-discussion] security testing In-Reply-To: <147034.3651.qm@web37506.mail.mud.yahoo.com> Message-ID: Apologies for the shameless plug. My company makes the Mu-4000 Security Analyzer that (among other things) has about 2,000,000 deeply stateful test cases designed to expose protocol implementation flaws. We can deliver those 2 million test cases over any of 5 transports (UDP, TCP, SSLv2, SSLv3, and TLSv1). Early next year, all those will work over IPv6, making 10 different "transport stacks" for SIP. We also support some IMS options that can affect our SIP test cases. The analyzer is highly automated and can monitor the target any way you like. The analyzer also does response-time profiles for how the invalid test cases we are sending affects the target's ability to respond to valid traffic. We have participated in SIPit events for the last year or so (see you in Beijing?) and after next week will have participated in all three IMS Plugfests (so far!). CT Labs uses the product as well, in their VoIP testing facility. Cheers, ~tom On 10/10/07 4:06 AM, "nazeema Tasneem" wrote: > Hi all, > can anyone tell me about automated tools for testing > security considerations(message flooding, registration > hijak, call teardown etc) in SIP. > > Thanks > Nazeema From raosiponline at gmail.com Thu Oct 11 05:51:16 2007 From: raosiponline at gmail.com (sambasivarao Vemula) Date: Thu, 11 Oct 2007 11:21:16 +0530 Subject: [SIPForum-discussion] security testing In-Reply-To: <147034.3651.qm@web37506.mail.mud.yahoo.com> References: <147034.3651.qm@web37506.mail.mud.yahoo.com> Message-ID: HI, Automation for registration flooding and registration hijak one tool is there i.e SIPPY tool ....is it available for open source or not ,I dont have any idea. Regards Samba On 10/10/07, nazeema Tasneem wrote: > > Hi all, > can anyone tell me about automated tools for testing > security considerations(message flooding, registration > hijak, call teardown etc) in SIP. > > Thanks > Nazeema > > > > > > ____________________________________________________________________________________ > Building a website is a piece of cake. Yahoo! Small Business gives you all > the tools to get online. > http://smallbusiness.yahoo.com/webhosting > _______________________________________________ > This is the SIP Forum discussion mailing list > TO UNSUBSCRIBE, or edit your delivery options, please visit > http://sipforum.org/mailman/listinfo/discussion > Post to the list at discussion at sipforum.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From VPFR47 at motorola.com Thu Oct 11 08:03:35 2007 From: VPFR47 at motorola.com (S Selvakumar-VPFR47) Date: Thu, 11 Oct 2007 16:03:35 +0800 Subject: [SIPForum-discussion] Replace header in REFER message Message-ID: <40E89886C8B3B54B98C5291646C591AA01BA29E5@ZMY16EXM67.ds.mot.com> Hi, Does anyone know on what basis the Replace header in REFER is build regards Selva -------------- next part -------------- An HTML attachment was scrubbed... URL: From chauhan_delhi at yahoo.com Thu Oct 11 08:40:09 2007 From: chauhan_delhi at yahoo.com (Ramesh Chauhan) Date: Thu, 11 Oct 2007 01:40:09 -0700 (PDT) Subject: [SIPForum-discussion] Extension.conf Message-ID: <332532.78809.qm@web34405.mail.mud.yahoo.com> Hi all, Entery in file is as folllows: SIP.CONF: [7795] type=friend username=7795 secret=7795 host=dynamic port=5060 relaxdtmf=yes dtmfmode=inband canreinvite=no context=outgoing_ISD mailbox=7795 at default disallow=all allow=ulaw EXTENSIONS.CONF: [outgoing_STD] exten => _0Zxxxxxxxxx,2,Dial(Zap/r1/${EXTEN}) exten => _77XX,1,Answer() exten => _77XX,2,Dial(SIP/${EXTEN},30,r) exten => _77XX,3,Hangup() Question: We want, if someome dials any outside number, it will ask for passwd. How to configure extension or any other file in that case ? PLease help me out.... Regards Chauhan --------------------------------- Shape Yahoo! in your own image. Join our Network Research Panel today! -------------- next part -------------- An HTML attachment was scrubbed... URL: From Karthik_Ramiya at infosys.com Thu Oct 11 09:26:40 2007 From: Karthik_Ramiya at infosys.com (Karthik Ramiya) Date: Thu, 11 Oct 2007 14:56:40 +0530 Subject: [SIPForum-discussion] Replace header in REFER message In-Reply-To: <40E89886C8B3B54B98C5291646C591AA01BA29E5@ZMY16EXM67.ds.mot.com> References: <40E89886C8B3B54B98C5291646C591AA01BA29E5@ZMY16EXM67.ds.mot.com> Message-ID: <13F04D2767AA2D42878E8ED4856BCB6502B0FA918E@BLRKECMBX02.ad.infosys.com> Hi Selva, Hope RFC3891 helps u. Thanks and regards, Karthik ________________________________ From: discussion-bounces at sipforum.org [mailto:discussion-bounces at sipforum.org] On Behalf Of S Selvakumar-VPFR47 Sent: Thursday, October 11, 2007 1:34 PM To: discussion at sipforum.org Subject: [SIPForum-discussion] Replace header in REFER message Hi, Does anyone know on what basis the Replace header in REFER is build regards Selva **************** CAUTION - Disclaimer ***************** This e-mail contains PRIVILEGED AND CONFIDENTIAL INFORMATION intended solely for the use of the addressee(s). If you are not the intended recipient, please notify the sender by e-mail and delete the original message. Further, you are not to copy, disclose, or distribute this e-mail or its contents to any other person and any such actions are unlawful. This e-mail may contain viruses. Infosys has taken every reasonable precaution to minimize this risk, but is not liable for any damage you may sustain as a result of any virus in this e-mail. You should carry out your own virus checks before opening the e-mail or attachment. Infosys reserves the right to monitor and review the content of all messages sent to or from this e-mail address. Messages sent to or from this e-mail address may be stored on the Infosys e-mail system. ***INFOSYS******** End of Disclaimer ********INFOSYS*** -------------- next part -------------- An HTML attachment was scrubbed... URL: From joel.silva at novabase.pt Thu Oct 11 10:28:24 2007 From: joel.silva at novabase.pt (Joel Silva) Date: Thu, 11 Oct 2007 11:28:24 +0100 Subject: [SIPForum-discussion] Edit sip-communicator layout References: <332532.78809.qm@web34405.mail.mud.yahoo.com> Message-ID: Hello. I would like to edit the layout of the communicator but I find it to complex. Can someone give me some help or advise me some tools to do this? Thanks, Joel. ________________________________ From: discussion-bounces at sipforum.org [mailto:discussion-bounces at sipforum.org] On Behalf Of Ramesh Chauhan Sent: quinta-feira, 11 de Outubro de 2007 9:40 To: discussion at sipforum.org Subject: [SPAM] - [SIPForum-discussion] Extension.conf - Sending mail server found on xbl.spamhaus.org Hi all, Entery in file is as folllows: SIP.CONF: [7795] type=friend username=7795 secret=7795 host=dynamic port=5060 relaxdtmf=yes dtmfmode=inband canreinvite=no context=outgoing_ISD mailbox=7795 at default disallow=all allow=ulaw EXTENSIONS.CONF: [outgoing_STD] exten => _0Zxxxxxxxxx,2,Dial(Zap/r1/${EXTEN}) exten => _77XX,1,Answer() exten => _77XX,2,Dial(SIP/${EXTEN},30,r) exten => _77XX,3,Hangup() Question: We want, if someome dials any outside number, it will ask for passwd. How to configure extension or any other file in that case ? PLease help me out.... Regards Chauhan ________________________________ Shape Yahoo! in your own image. Join our Network Research Panel today! -------------- next part -------------- An HTML attachment was scrubbed... URL: From deepanshu at huawei.com Fri Oct 12 01:35:26 2007 From: deepanshu at huawei.com (Deepanshu) Date: Fri, 12 Oct 2007 09:35:26 +0800 Subject: [SIPForum-discussion] Replace header in REFER message References: <40E89886C8B3B54B98C5291646C591AA01BA29E5@ZMY16EXM67.ds.mot.com> Message-ID: <005d01c80c70$2ab032c0$9e78a40a@china.huawei.com> Replace header is based on the dialog information (ID, to-tag and from-tag) of a current dialog. An INVITE with the Replace header will replace the current dialog. For example A is in dialog with B. C send a INVITE with replace (with dialog information) header to A. Dialog between A and B gets replaced by a new dialog between A and C. BR Deepanshu Gautam Huawei Technologies Co. Ltd. ----- Original Message ----- From: S Selvakumar-VPFR47 To: discussion at sipforum.org Sent: Thursday, October 11, 2007 4:03 PM Subject: [SIPForum-discussion] Replace header in REFER message Hi, Does anyone know on what basis the Replace header in REFER is build regards Selva ------------------------------------------------------------------------------ _______________________________________________ This is the SIP Forum discussion mailing list TO UNSUBSCRIBE, or edit your delivery options, please visit http://sipforum.org/mailman/listinfo/discussion Post to the list at discussion at sipforum.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From jwwlhc at yahoo.com.cn Fri Oct 12 03:19:38 2007 From: jwwlhc at yahoo.com.cn (=?gb2312?q?=CE=AA=20=BD=AA?=) Date: Fri, 12 Oct 2007 11:19:38 +0800 (CST) Subject: [SIPForum-discussion] (no subject) Message-ID: <368464.13422.qm@web15206.mail.cnb.yahoo.com> thanks! --------------------------------- @yahoo.cn ????????????? -------------- next part -------------- An HTML attachment was scrubbed... URL: From alexzhang at gdnt.com.cn Fri Oct 12 08:38:12 2007 From: alexzhang at gdnt.com.cn (alexzhang at gdnt.com.cn) Date: Fri, 12 Oct 2007 16:38:12 +0800 Subject: [SIPForum-discussion] Is the non-2xx final response before the PRACK of the previous provisonal response allowed? Message-ID: <8E523FC208B8174790E69947E307914702083C82@rnd-ex01.rnd.gdnt.local> Hi, Here I am wondering if this scenario can be compliant the SIP protocol? UAC UAS --------- Invite --------------> <--------100 Trying--------- <---------183 SP ----------- something happened before the PRACK <---------4xx failure-------- Alex Zhang Guangdong Nortel (GDNT) R&D Center GSM/UMTS Voice Core - MSC Design Phone: (PSTN)+86 020 89188782 (ESN)6 554 8782 E-mail: alexzhang at gdnt.com.cn YahooIM: zcc_nuaa -------------- next part -------------- An HTML attachment was scrubbed... URL: From avorlando at yahoo.com Fri Oct 12 12:53:55 2007 From: avorlando at yahoo.com (Anthony Orlando) Date: Fri, 12 Oct 2007 05:53:55 -0700 (PDT) Subject: [SIPForum-discussion] Request timeouts Message-ID: <423261.67192.qm@web51010.mail.re2.yahoo.com> All, I have a question that I can't find an answer for in any RFC etc. My question is as follows. I have a S-CSCF that has done a DNS query for an application server pair that runs in an active standby mode. I am returned an address for two application servers with priority 1 for primary and 100 for secondary. A call is established on AS1 then is failed. AS2 becomes active. When the phone is hung up it sends BYE messages from the S-CSCF to AS1 for five minutes before he tries the secondary. Obviously this is too long. Can anyone reference an RFC that covers this? Thoughts or feelings are also welcome. ____________________________________________________________________________________ Moody friends. Drama queens. Your life? Nope! - their life, your story. Play Sims Stories at Yahoo! Games. http://sims.yahoo.com/ From marc.robins at sipforum.org Fri Oct 12 16:12:31 2007 From: marc.robins at sipforum.org (Marc Robins) Date: Fri, 12 Oct 2007 12:12:31 -0400 Subject: [SIPForum-discussion] The IIT VoIP Conference and EXPO 2007 Message-ID: <005601c80cea$b17f2700$6501a8c0@RCG> Dear SIP Forum Members, The SIP Forum is proud to announce that we are co-sponsoring the upcoming third annual VoIP Conference and EXPO, hosted by the Illinois Institute of Technology in Wheaton, Illinois. Taking place on Thursday and Friday, October 25 and 26, 2007, VoIP Conference and EXPO 2007 features more than 38 sessions from industry experts and a wine and cheese networking event. The full conference agenda is available at http://www.cpd.iit.edu/voipconference07/schedule.php. Registration Information: As part of the Forum?s co-sponsorship, individual Forum members are entitled to attend the conference at the group rate of $125. Conference Registration includes admission to both days of the conference and expo, breakfast and lunch on both days, the wine and cheese reception on Thursday evening, presenter materials and conference tote bag. To register for the event, please visit http://www.cpd.iit.edu/voipconference07/register.php Participating Organizations: Participating organizations include: Alcatel-Lucent, AT&T, Acme Packet, Azaire Networks, BIT, Booz-Allen-Hamilton, Cbeyond, Cimco, Columbia University, Digium-Asterisk, Enabling Technologies, GeckoTech, Informity Networks, IEEE ComSoc, IIT, International Engineering Consortium, Intrado, M5, Morgan Franklin, Motorola, Neustar, Occam, Penn State University, Performance Technologies, Reef Point Systems, SIP Forum, Spirent, Telchemy, Telcordia, Teleprime, YS-41 Communications, Westell and many others. Exhibit/Sponsorship Opportunities: There are still opportunities to exhibit and sponsor the event. Rates include: Gold Sponsor & Exhibitor - $750 (2 free attendees) Silver Sponsor & Exhibitor - $550 (2 free attendees) Exhibitor only - $325 (2 free attendees) Sponsor only - $300 (1 free attendee) For full details, visit http://www.cpd.iit.edu/voipconference07/exhibitorsponsor.php For More Information: Complete information about the conference can be found at the conference website at http://www.cpd.iit.edu/voipconference07. For questions, contact Scott Pfeiffer at pfeiffer at iit.edu or by phone at 630-682-6001. ************************* Marc Robins Managing Director Elect, SIP Forum www.sipforum.org Chief Evangelism Officer, RCG Tel: 718-548-7245 Mobile: 203-829-6307 SkypeMe! marcrobins http://www.robinsconsult.com ************************* -------------- next part -------------- An HTML attachment was scrubbed... URL: From suba_technical at yahoo.co.in Sat Oct 13 10:42:03 2007 From: suba_technical at yahoo.co.in (Subashini Rajaraman) Date: Sat, 13 Oct 2007 11:42:03 +0100 (BST) Subject: [SIPForum-discussion] Fax-Passthrough in AC48301! Message-ID: <170161.51455.qm@web94415.mail.in2.yahoo.com> Hi All, In fax-pass through What is the rtp payload type in AC48301. Wheather this Dsp supports all the four types of mode. Fax relay Fax passthrough Modem Relay Modem passthrough I guess the Rtp payload type should be the user configured coder or high bit coder(G711) in Fax passthrough mode. Is it Correct? How the Dsp AC48301 detects the CED tone in Fax passthrough mode.what is the modem rate should be used.Can we relay enable any of the modem type in the fax pass through mode. I had a problem in detecting CED tone in AC48301.According to the manual what are the modems should be relayenable and bypass enable in Faxpassthrough mode & why?. If all should be bypassenable then how the Fax messages(CED,training)will be detected by the DSP. --- Thanks in advance, subashini. --------------------------------- Forgot the famous last words? Access your message archive online. Click here. -------------- next part -------------- An HTML attachment was scrubbed... URL: From mohammadfaiz2003 at yahoo.com Sun Oct 14 09:01:51 2007 From: mohammadfaiz2003 at yahoo.com (�The Passenger�) Date: Sun, 14 Oct 2007 02:01:51 -0700 (PDT) Subject: [SIPForum-discussion] (no subject) Message-ID: <537097.29250.qm@web39611.mail.mud.yahoo.com> mohammadfaiz2003 at yahoo.com ((mOhAmAd fAiZ)) --------------------------------- Moody friends. Drama queens. Your life? Nope! - their life, your story. Play Sims Stories at Yahoo! Games. -------------- next part -------------- An HTML attachment was scrubbed... URL: From mohammadfaiz2003 at yahoo.com Sun Oct 14 09:19:19 2007 From: mohammadfaiz2003 at yahoo.com (�The Passenger�) Date: Sun, 14 Oct 2007 02:19:19 -0700 (PDT) Subject: [SIPForum-discussion] Questions Message-ID: <185264.68570.qm@web39612.mail.mud.yahoo.com> Hi every one I'm a new member in SIP Forum and I?m master student from Malaysia I?m doing my master in implementation between SIP and another multimedia protocol and I just want to know some issues 1. What is the codec that SIP supports them? 2. Are RTP and RSTP the only protocols that SIP uses them for exchanging media? 3. I know that I need to do signaling translation and media translation as well, so what are the important issues that I need to put it in my consideration? 4. If you have any other information that you think it important and I need to know it please I will appreciate that Best Regards ((mOhAmAd fAiZ)) --------------------------------- Need a vacation? Get great deals to amazing places on Yahoo! Travel. -------------- next part -------------- An HTML attachment was scrubbed... URL: From pinakee.b at xius.com Mon Oct 15 04:12:34 2007 From: pinakee.b at xius.com (Pinakeeb) Date: Mon, 15 Oct 2007 09:42:34 +0530 Subject: [SIPForum-discussion] Request timeouts In-Reply-To: <423261.67192.qm@web51010.mail.re2.yahoo.com> Message-ID: <200710150412.l9F4CC3h022420@serv1.xius.com> Anthony, The scenario mentioned in your mail is from IMS. I don't think there is any RFC for IMS. IMS is defined in 3GPP. You will find the standards there - http://www.3gpp.org Cheers, Pinakee -----Original Message----- From: discussion-bounces at sipforum.org [mailto:discussion-bounces at sipforum.org] On Behalf Of Anthony Orlando Sent: Friday, October 12, 2007 6:24 PM To: discussion at sipforum.org Subject: [SIPForum-discussion] Request timeouts All, I have a question that I can't find an answer for in any RFC etc. My question is as follows. I have a S-CSCF that has done a DNS query for an application server pair that runs in an active standby mode. I am returned an address for two application servers with priority 1 for primary and 100 for secondary. A call is established on AS1 then is failed. AS2 becomes active. When the phone is hung up it sends BYE messages from the S-CSCF to AS1 for five minutes before he tries the secondary. Obviously this is too long. Can anyone reference an RFC that covers this? Thoughts or feelings are also welcome. ____________________________________________________________________________ ________ Moody friends. Drama queens. Your life? Nope! - their life, your story. Play Sims Stories at Yahoo! Games. http://sims.yahoo.com/ _______________________________________________ This is the SIP Forum discussion mailing list TO UNSUBSCRIBE, or edit your delivery options, please visit http://sipforum.org/mailman/listinfo/discussion Post to the list at discussion at sipforum.org From adams-gao at hotmail.com Mon Oct 15 06:54:37 2007 From: adams-gao at hotmail.com (=?gb2312?B?uN+9qA==?=) Date: Mon, 15 Oct 2007 14:54:37 +0800 Subject: [SIPForum-discussion] "Failed to bind to socket 1868" Message-ID: Hi,All: I am a fresh man of using SipXtapi,there are some problems when i build my project with c#. My Server using Asterisk,and X-lite works on it well,but the sipXezphone i complied does not work, it regists failed. i used sipXezphone default settings,it always returns "Failed to bind to socket 1868". and the log writes as fellows,any one can give me a hand? with many thanks! "2007-10-15T06:26:10.949000Z":3:SIPXTAPI:DEBUG:e-5d2790a721fd4::00000000:sipXtapi:"ENTER FUNC (tid=3744) sipxInitialize\n""2007-10-15T06:26:10.949000Z":4:SIPXTAPI:DEBUG:e-5d2790a721fd4::00000000:sipXtapi:"ENTER FUNC (tid=3744) sipxConfigGetVersion\n""2007-10-15T06:26:10.949000Z":5:SIPXTAPI:DEBUG:e-5d2790a721fd4::00000000:sipXtapi:"EXIT FUNC (tid=3744) sipxConfigGetVersion\n""2007-10-15T06:26:10.949000Z":6:SIPXTAPI:INFO:e-5d2790a721fd4::00000000:sipXtapi:"sipXtapi SDK 2.9.1.0 Dbg (built 0000-00-00)""2007-10-15T06:26:10.965000Z":7:SIPXTAPI:INFO:e-5d2790a721fd4::00000000:sipXtapi:"sipxInitialize tcpPort=36004 udpPort=36004 tlsPort=0 rtpPortStart=9000 maxConnections=32 identity=sipx bindTo=0.0.0.0 sequentialPorts=0 certNickname=800, DBLocation=""2007-10-15T06:26:10.965000Z":8:SIPXTAPI:DEBUG:e-5d2790a721fd4::00000000:sipXtapi:"ENTER FUNC (tid=3744) sipxConfigGetAllLocalNetworkIps\n""2007-10-15T06:26:10.965000Z":9:SIPXTAPI:INFO:e-5d2790a721fd4::00000000:sipXtapi:"sipxConfigGetAllLocalNetworkIps index=0 address=168.150.10.241 adapter=eth0""2007-10-15T06:26:10.965000Z":10:SIPXTAPI:DEBUG:e-5d2790a721fd4::00000000:sipXtapi:"EXIT FUNC (tid=3744) sipxConfigGetAllLocalNetworkIps\n""2007-10-15T06:26:10.965000Z":11:SIPXTAPI:INFO:e-5d2790a721fd4::00000000:sipXtapi:"sipXtapi built without NSS support""2007-10-15T06:26:10.965000Z":12:SIP:DEBUG:e-5d2790a721fd4::00000000:sipXtapi:"SipUserAgent::_ sipTcpPort = 36004, sipUdpPort = 36004, sipTlsPort = 0""2007-10-15T06:26:10.965000Z":13:SIP:DEBUG:e-5d2790a721fd4::00000000:sipXtapi:"SipUdpServer::_ port = 36004, bUseNextAvailablePort = 0, szBoundIp = '0.0.0.0'""2007-10-15T06:26:10.980000Z":14:KERNEL:ERR:e-5d2790a721fd4::00000000:sipXtapi:"Failed to bind to socket 1868\n""2007-10-15T06:26:10.980000Z":15:SIP:DEBUG:e-5d2790a721fd4::00000000:sipXtapi:"SipTcpServer::_ port = 36004, taskName = 'SipTcpServer-%d', bUseNextAvailablePort = 0, szBindAddr = '0.0.0.0'""2007-10-15T06:26:10.980000Z":16:KERNEL:DEBUG:e-5d2790a721fd4::00000000:sipXtapi:"OsServerSocket::_ queue=64 port=36004 bindaddr=16 8.150.10.241""2007-10-15T06:26:10.980000Z":17:KERNEL:ERR:e-5d2790a721fd4::00000000:sipXtapi:"OsServerSocket: bind to port 36004 failed with error: 10048 = 0x2740""2007-10-15T06:26:10.996000Z":18:KERNEL:DEBUG:e-5d2790a721fd4::00000000:sipXtapi:"OsTimerTask::getTimerTask OsTimerTask started""2007-10-15T06:26:10.996000Z":19:SIPXTAPI:INFO:e-5d2790a721fd4::00000000:sipXtapi:"Default bind address 168.150.10.241, udpPort=36004, tcpPort=36004, tlsPort=-1""2007-10-15T06:26:10.996000Z":20:SIPXTAPI:INFO:e-5d2790a721fd4::00000000:sipXtapi:"Default Identity: sip:sipx at 168.150.10.241:36004\n""2007-10-15T06:26:11.168000Z":21:KERNEL:DEBUG:e-5d2790a721fd4:NetInTask:000006B0:sipXtapi:"OsServerSocket::_ queue=1 port=-2 bindaddr=127.0.0.1""2007-10-15T06:26:11.168000Z":22:SIP:DEBUG:e-5d2790a721fd4:NetInTaskHelper-15:00000DC0:sipXtapi:"OsConnectionSocket::_ attempt 127.0.0.1:2438 BLOCKING""2007-10-15T06:26:11.168000Z":23:MP:INFO:e-5d2790a721fd4:NetInTaskHelper-15:FFFFFFFF:sipXtapi:"NetInTaskHelper::run()... returning 0, after 1 tries\n""2007-10-15T06:26:11.277000Z":26:CP:DEBUG:e-5d2790a721fd4::00000000:sipXtapi:"ENTER FUNC (tid=3744) CallManager\n""2007-10-15T06:26:11.277000Z":27:CP:DEBUG:e-5d2790a721fd4::00000000:sipXtapi:"EXIT FUNC (tid=3744) CallManager\n""2007-10-15T06:26:11.277000Z":28:SIPXTAPI:DEBUG:e-5d2790a721fd4::00000000:sipXtapi:"ENTER FUNC (tid=3744) sipxConfigSetAudioCodecPreferences\n""2007-10-15T06:26:11.277000Z":29:SIPXTAPI:INFO:e-5d2790a721fd4::00000000:sipXtapi:"sipxConfigSetAudioCodecPreferences hInst=04A45830 bandWidth=2""2007-10-15T06:26:11.277000Z":30:SIPXTAPI:DEBUG:e-5d2790a721fd4::00000000:sipXtapi:"sipxConfigSetAudioCodecPreferences number of Codec = 8 for hInst=04A45830""2007-10-15T06:26:11.277000Z":31:SIPXTAPI:DEBUG:e-5d2790a721fd4::00000000:sipXtapi:"sipxConfigSetAudioCodecPreferences: PCMU PCMA audio/telephone-event SPEEX SPEEX_5 SPEEX_15 SPEEX_24 GSM""2007-10-15T06:26:11.277000Z":32:MP:DEBUG:e-5d2790a721fd4::00000000:sipXtapi:"sipXmediaFactoryImpl::buildCodecFactory: sReferences = PCMU PCMA audio/tele phone-event SPEEX SPEEX_5 SPEEX_15 SPEEX_24 GSM with NumReject 0""2007-10-15T06:26:11.277000Z":33:MP:DEBUG:e-5d2790a721fd4::00000000:sipXtapi:"sipXmediaFactoryImpl::buildCodecFactory: supported codecs = PCMU PCMA audio/telephone-event SPEEX SPEEX_5 SPEEX_15 SPEEX_24 GSM with NumReject 0""2007-10-15T06:26:11.277000Z":34:SIPXTAPI:DEBUG:e-5d2790a721fd4::00000000:sipXtapi:"EXIT FUNC (tid=3744) sipxConfigSetAudioCodecPreferences\n""2007-10-15T06:26:11.277000Z":35:SIPXTAPI:INFO:e-5d2790a721fd4::00000000:sipXtapi:"__sipxEventListenerAdd hInst=04A45830 pCallbackProc=047E1810 pUserData=04ACE618""2007-10-15T06:26:11.777000Z":36:SIPXTAPI:DEBUG:e-5d2790a721fd4::00000000:sipXtapi:"EXIT FUNC (tid=3744) sipxInitialize\n" _________________________________________________________________ Windows Live Spaces ???????? 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URL: From alexzhang at gdnt.com.cn Tue Oct 16 03:08:20 2007 From: alexzhang at gdnt.com.cn (alexzhang at gdnt.com.cn) Date: Tue, 16 Oct 2007 11:08:20 +0800 Subject: [SIPForum-discussion] [Sip-implementors] The non-2xx final response is allowed to besentby UAS before PRACK of the Provisional Response? In-Reply-To: <8E523FC208B8174790E69947E3079147020842D6@rnd-ex01.rnd.gdnt.local> References: <8E523FC208B8174790E69947E307914702083F1B@rnd-ex01.rnd.gdnt.local><0D4E483A0E6E0A46861409E5D6C2011CCF5DFA@sea02-mxc01.citel.com> <8E523FC208B8174790E69947E3079147020842D6@rnd-ex01.rnd.gdnt.local> Message-ID: <8E523FC208B8174790E69947E30791470208432C@rnd-ex01.rnd.gdnt.local> Or does the spec allow the below scenario? UAC UAS |-------Inivte(SDP Offer)--------------->| |<------100 Trying-----------------------| | | |<------183 Session Prog(SDP Anser)------| | | |<------480 Temp Fail--------------------| | | Thanks, Alex 6-554-8782 -----Original Message----- From: sip-implementors-bounces at lists.cs.columbia.edu [mailto:sip-implementors-bounces at lists.cs.columbia.edu] On Behalf Of alexzhang at gdnt.com.cn Sent: Tuesday, October 16, 2007 10:18 AM To: michael.procter at citel.com; sip-implementors at lists.cs.columbia.edu Subject: Re: [Sip-implementors] The non-2xx final response is allowed to besentby UAS before PRACK of the Provisional Response? I am a little about the sentence: " unless the final response is 2xx and any of the unacknowledged reliable provisional responses contained a session description." Is it meaning that if the prvisional response contains the SDP, the UAS must wait for the PRACK before sending the final response? Or, if the provisional response contains the SDP, the UAS must wait for the PRACK before sending the 2xx final response? Thanks, Alex 6-554-8782 -----Original Message----- From: Michael Procter [mailto:michael.procter at citel.com] Sent: Monday, October 15, 2007 4:26 PM To: Alex Zhang (GDNTRND); sip-implementors at lists.cs.columbia.edu Subject: RE: [Sip-implementors] The non-2xx final response is allowed to be sentby UAS before PRACK of the Provisional Response? Yes. RFC 3262, section 3, paragraph 19 (last paragraph in section): The UAS MAY send a final response to the initial request before having received PRACKs for all unacknowledged reliable provisional responses, unless the final response is 2xx and any of the unacknowledged reliable provisional responses contained a session description. In that case, it MUST NOT send a final response until those provisional responses are acknowledged. If the UAS does send a final response when reliable responses are still unacknowledged, it SHOULD NOT continue to retransmit the unacknowledged reliable provisional responses, but it MUST be prepared to process PRACK requests for those outstanding responses. A UAS MUST NOT send new reliable provisional responses (as opposed to retransmissions of unacknowledged ones) after sending a final response to a request. Regards, Michael > -----Original Message----- > From: sip-implementors-bounces at lists.cs.columbia.edu [mailto:sip- > implementors-bounces at lists.cs.columbia.edu] On Behalf Of > alexzhang at gdnt.com.cn > Sent: 15 October 2007 04:15 > To: sip-implementors at lists.cs.columbia.edu > Subject: [Sip-implementors] The non-2xx final response is allowed to > be sentby UAS before PRACK of the Provisional Response? > > Hi, Here I am wondering if this scenario can be compliant to the SIP > protocol? > > UAC UAS > --------- Invite --------------> > <--------100 Trying--------- > <---------183 SP ----------- > something happened > before the PRACK > <---------4xx failure-------- > > > > > Alex Zhang > Guangdong Nortel (GDNT) R&D Center > GSM/UMTS Voice Core - MSC Design > Phone: (PSTN)+86 020 89188782 (ESN)6 554 8782 > E-mail: alexzhang at gdnt.com.cn > YahooIM: zcc_nuaa > > > > > > Alex Zhang > Guangdong Nortel (GDNT) R&D Center > GSM/UMTS Voice Core - MSC Design > Phone: (PSTN)+86 020 89188782 (ESN)6 554 8782 > E-mail: alexzhang at gdnt.com.cn YahooIM: > zcc_nuaa > > > _______________________________________________ > Sip-implementors mailing list > Sip-implementors at lists.cs.columbia.edu > https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors _______________________________________________ Sip-implementors mailing list Sip-implementors at lists.cs.columbia.edu https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors From adams-gao at hotmail.com Thu Oct 18 03:41:24 2007 From: adams-gao at hotmail.com (=?gb2312?B?uN+9qA==?=) Date: Thu, 18 Oct 2007 11:41:24 +0800 Subject: [SIPForum-discussion] sipxezphone register time out Message-ID: Hi,All when i using sipxezphone send register message,the server(asterisk)always returns register failure infomation 401,and the UC seems dosent response it,and it always send register message but no re-register with authorization. refering about the sip pack by Ethereal,i found the UC 's souce port was 2755,and distination port was 5060(default),but the response(trying and unauthorize) distination port was 5060. so i guess the port maybe not correct,and the UC cannt recive response message? can anyone give me a hand and tell me how to solute this problem? thanks!Adams.Gao _________________________________________________________________ MSN ???????????????????? http://cn.msn.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From Xiaohong.Xiao at alcatel-sbell.com.cn Thu Oct 18 05:24:23 2007 From: Xiaohong.Xiao at alcatel-sbell.com.cn (XIAO Xiaohong) Date: Thu, 18 Oct 2007 13:24:23 +0800 Subject: [SIPForum-discussion] what's the difference & relations between SIP-I and SIP-T? In-Reply-To: References: Message-ID: -------------- next part -------------- An HTML attachment was scrubbed... URL: From zhousiru at gmail.com Thu Oct 18 05:42:12 2007 From: zhousiru at gmail.com (siru zhou) Date: Thu, 18 Oct 2007 13:42:12 +0800 Subject: [SIPForum-discussion] sipxezphone register time out In-Reply-To: References: Message-ID: hi, gao 1. a test using another uac (e.g. Xlite) to see if there's something wrong with your sipxezphone congifguration. 2. is the issue related to NAT ? 3. dump and paste the detail of session packages here BR Siru On 10/18/07, ?? wrote: > > Hi,All > when i using sipxezphone send register message,the > server(asterisk)always returns register failure infomation 401,and the UC > seems dosent response it,and it always send register message but no > re-register with authorization. > refering about the sip pack by Ethereal,i found the UC 's souce port > was 2755,and distination port was 5060(default),but the response(trying and > unauthorize) distination port was 5060. > so i guess the port maybe not correct,and the UC cannt recive response > message? > can anyone give me a hand and tell me how to solute this problem? > > thanks! > Adams.Gao > > ------------------------------ > ???????????????? Windows Live Mail? ????? > > _______________________________________________ > This is the SIP Forum discussion mailing list > TO UNSUBSCRIBE, or edit your delivery options, please visit > http://sipforum.org/mailman/listinfo/discussion > Post to the list at discussion at sipforum.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From ytian at juniper.net Thu Oct 18 05:47:32 2007 From: ytian at juniper.net (Yong Tian) Date: Thu, 18 Oct 2007 13:47:32 +0800 Subject: [SIPForum-discussion] sipxezphone register time out In-Reply-To: Message-ID: <7B8EBFC47BB8C24F80FF697FC6C7B6DFAA9A54@emailcnrd1.jnpr.net> Hi Adams, What is the VIA header in your request? Could you check? YOng ________________________________ From: discussion-bounces at sipforum.org [mailto:discussion-bounces at sipforum.org] On Behalf Of ?? Sent: 2007?10?18? 11:41 To: discussion at sipforum.org Subject: [SIPForum-discussion] sipxezphone register time out Hi,All when i using sipxezphone send register message,the server(asterisk)always returns register failure infomation 401,and the UC seems dosent response it,and it always send register message but no re-register with authorization. refering about the sip pack by Ethereal,i found the UC 's souce port was 2755,and distination port was 5060(default),but the response(trying and unauthorize) distination port was 5060. so i guess the port maybe not correct,and the UC cannt recive response message? can anyone give me a hand and tell me how to solute this problem? thanks! Adams.Gao ________________________________ ???????????????? Windows Live Mail? ????? -------------- next part -------------- An HTML attachment was scrubbed... URL: From jnm_04 at rediffmail.com Thu Oct 18 05:51:56 2007 From: jnm_04 at rediffmail.com (jitendra mohapatra) Date: 18 Oct 2007 05:51:56 -0000 Subject: [SIPForum-discussion] How many dialoge, transaction and call are created Message-ID: <20071018055156.20747.qmail@f5mail12.rediffmail.com> HI everyone I am jitendra newly started SIP kindle anyone tell me how many call ,transaction and dialog created in case of call transfer-attended regards jitendra.. -------------- next part -------------- An HTML attachment was scrubbed... URL: From kunusan at yahoo.com Thu Oct 18 06:05:01 2007 From: kunusan at yahoo.com (badal naik) Date: Wed, 17 Oct 2007 23:05:01 -0700 (PDT) Subject: [SIPForum-discussion] what's the difference & relations between SIP-I and SIP-T? In-Reply-To: Message-ID: <258184.17290.qm@web56706.mail.re3.yahoo.com> SIP-T stands a general framework for internetworking with ISUP.It sees Sip to carry ISUP as MIME Body. SIP-I:Provides more details about how encapsulation and mapping are to be performed at NNI Interface. Example:when De-encaspulating ISUP message: SIP-T: Sees it as a template overriden by SIP headers. SIP-I:Give details about what information should be taken from ISUP and what to retrive from internetworking fro SIP headers to ISUP parametres. Thanks Badal --- XIAO Xiaohong wrote: > > > _______________________________________________ > This is the SIP Forum discussion mailing list > TO UNSUBSCRIBE, or edit your delivery options, > please visit > http://sipforum.org/mailman/listinfo/discussion > Post to the list at discussion at sipforum.org > ____________________________________________________________________________________ Boardwalk for $500? In 2007? Ha! Play Monopoly Here and Now (it's updated for today's economy) at Yahoo! Games. http://get.games.yahoo.com/proddesc?gamekey=monopolyherenow From sukerry at 126.com Thu Oct 18 06:57:44 2007 From: sukerry at 126.com (Kerry) Date: Thu, 18 Oct 2007 14:57:44 +0800 Subject: [SIPForum-discussion] sipxezphone register time out Message-ID: <471703F0.08A0FC.02526@m15-112.126.com> siru zhou???? ??please try to set realm fied as "asterisk" ======== 2007-10-18 13:42:12 ???????? ======== hi, gao 1. a test using another uac (e.g. Xlite) to see if there's something wrong with your sipxezphone congifguration. 2. is the issue related to NAT ? 3. dump and paste the detail of session packages here BR Siru On 10/18/07, ?? wrote: Hi,All when i using sipxezphone send register message,the server(asterisk)always returns register failure infomation 401,and the UC seems dosent response it,and it always send register message but no re-register with authorization. refering about the sip pack by Ethereal,i found the UC 's souce port was 2755,and distination port was 5060(default),but the response(trying and unauthorize) distination port was 5060. so i guess the port maybe not correct,and the UC cannt recive response message? can anyone give me a hand and tell me how to solute this problem? thanks! Adams.Gao ???????????????? Windows Live Mail? ????? _______________________________________________ This is the SIP Forum discussion mailing list TO UNSUBSCRIBE, or edit your delivery options, please visit http://sipforum.org/mailman/listinfo/discussion Post to the list at discussion at sipforum.org = = = = = = = = = = = = = = = = = = = = = = ????????? ?? ??????????????Kerry ??????????????sukerry at 126.com ------------------------------------------------- ??????????????,??? Seawolf Technologies Co. Ltd. Beijing Tel: +86-10-82253150-611 (Office) MSN: suxiangmao at hotmail.com WebSite & Service: http://www.seawolftech.com/ http://www.xrainbow.com.cn/ http://www.17ip.com/ http://www.en400.com/ -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: seawolf_logo(09-20-11-42-22)1.ico Type: application/octet-stream Size: 9062 bytes Desc: not available URL: From mrknayak at gmail.com Thu Oct 18 13:29:55 2007 From: mrknayak at gmail.com (Rama krushna Nayak) Date: Thu, 18 Oct 2007 18:59:55 +0530 Subject: [SIPForum-discussion] 5 test case each on PRACK and Conference Message-ID: Hi All, Can anyone tell me 5 test case brifly on each. 1. using PRACK . 2.conference using 3 IP phone. Regards Ramakrushna -------------- next part -------------- An HTML attachment was scrubbed... URL: From mmostafa at nile-online.net Thu Oct 18 14:57:06 2007 From: mmostafa at nile-online.net (Mostafa Ali) Date: Thu, 18 Oct 2007 16:57:06 +0200 Subject: [SIPForum-discussion] Free CDR Server / Collector References: Message-ID: <026a01c81197$26793270$e2408c3e@engteam565> Dear All , Can anybody recommend a free CDR Serve . BR Mostafa Ali From joel.silva at novabase.pt Thu Oct 18 15:49:56 2007 From: joel.silva at novabase.pt (Joel Silva) Date: Thu, 18 Oct 2007 16:49:56 +0100 Subject: [SIPForum-discussion] Recommend sip/simple client sdk References: <026a01c81197$26793270$e2408c3e@engteam565> Message-ID: Can anyone recommend me a SDK/Simple client sdk? Preferably free, if not a trial version. Thanks, Joel From usman at my.web.pk Thu Oct 18 19:32:21 2007 From: usman at my.web.pk (Usman Rauf) Date: Fri, 19 Oct 2007 00:32:21 +0500 Subject: [SIPForum-discussion] Recommend sip/simple client sdk In-Reply-To: Message-ID: <200710181932.l9IJW0ag026144@sipforum.org> Hello, I need to implement SIP client and server in .Net. I have gone through the basics of SIP architecture but I am not getting a point to take a start in coding .. Can someone guide me on this plz? I'll be extremely thankful Regards, Usman Rauf. -----Original Message----- From: discussion-bounces at sipforum.org [mailto:discussion-bounces at sipforum.org] On Behalf Of Joel Silva Sent: Thursday, October 18, 2007 8:50 PM To: discussion at sipforum.org Subject: [SIPForum-discussion] Recommend sip/simple client sdk Can anyone recommend me a SDK/Simple client sdk? Preferably free, if not a trial version. Thanks, Joel _______________________________________________ This is the SIP Forum discussion mailing list TO UNSUBSCRIBE, or edit your delivery options, please visit http://sipforum.org/mailman/listinfo/discussion Post to the list at discussion at sipforum.org CONFIDENTIALITY NOTICE: This e-mail and its attachments (if any) contain information that is privileged, confidential and subject to legal restrictions and penalties regarding its unauthorized disclosure or other use. The information contained herein is the property of F3 Technologies. It is for the exclusive use of the intended recipient and may not be copied in any way, shape or form, or transmitted in any manner to any other distributor, individual, company or corporation. From mohammadfaiz2003 at yahoo.com Fri Oct 19 02:22:03 2007 From: mohammadfaiz2003 at yahoo.com (�The Passenger�) Date: Thu, 18 Oct 2007 19:22:03 -0700 (PDT) Subject: [SIPForum-discussion] CODEC Message-ID: <460087.92365.qm@web39607.mail.mud.yahoo.com> Hi every body is any one can tell me what is the codec which are already supported by SIP ? Audio and Video Codec also, i know its alot but please if you can memorize some of them thanks alot ((mOhAmAd fAiZ)) __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From yongjie.fan at gmail.com Fri Oct 19 02:49:10 2007 From: yongjie.fan at gmail.com (fan yongjie) Date: Fri, 19 Oct 2007 10:49:10 +0800 Subject: [SIPForum-discussion] Recommend sip/simple client sdk In-Reply-To: References: <026a01c81197$26793270$e2408c3e@engteam565> Message-ID: <73f4c8ed0710181949s1dc526c7l41d9da851d30fe0f@mail.gmail.com> you can find what you need from the link www.SIPfoundry.org 2007/10/18, Joel Silva : > > > Can anyone recommend me a SDK/Simple client sdk? Preferably free, if not > a trial version. > Thanks, > Joel > > _______________________________________________ > This is the SIP Forum discussion mailing list > TO UNSUBSCRIBE, or edit your delivery options, please visit > http://sipforum.org/mailman/listinfo/discussion > Post to the list at discussion at sipforum.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From yongjie.fan at gmail.com Fri Oct 19 03:03:19 2007 From: yongjie.fan at gmail.com (fan yongjie) Date: Fri, 19 Oct 2007 11:03:19 +0800 Subject: [SIPForum-discussion] CODEC In-Reply-To: <460087.92365.qm@web39607.mail.mud.yahoo.com> References: <460087.92365.qm@web39607.mail.mud.yahoo.com> Message-ID: <73f4c8ed0710182003tf7401dbxa9c297c62dd00718@mail.gmail.com> SIP does not support codecs directly. If you want to support some codecs, you can try to get some free project to implement it. for example, pjmedia the normal codecs include, G.711u/a, G.729, etc. 2007/10/19, ?The Passenger? : > > Hi every body is any one can tell me what is the codec which are already > supported by SIP ? Audio and Video Codec also, i know its alot but please > if you can memorize some of them > > thanks alot > > > *((mOhAmAd fAiZ))* > > __________________________________________________ > Do You Yahoo!? > Tired of spam? Yahoo! Mail has the best spam protection around > http://mail.yahoo.com > > _______________________________________________ > This is the SIP Forum discussion mailing list > TO UNSUBSCRIBE, or edit your delivery options, please visit > http://sipforum.org/mailman/listinfo/discussion > Post to the list at discussion at sipforum.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From Richard.Agonias at digitel.ph Fri Oct 19 03:35:28 2007 From: Richard.Agonias at digitel.ph (Agonias, Richard L. (Digitel-GSM)) Date: Fri, 19 Oct 2007 11:35:28 +0800 Subject: [SIPForum-discussion] SIP and SDP In-Reply-To: <73f4c8ed0710182003tf7401dbxa9c297c62dd00718@mail.gmail.com> Message-ID: <90B7FB18EBCD424B83BA9FEA0D4319E84095E0@dgtlmail.digitel.ph> Hi, What's the relation between SDP and SIP? Does anyone here have the complete layer stack? Thanks! -chad -------------- next part -------------- An HTML attachment was scrubbed... URL: From deveshbissa at rediffmail.com Fri Oct 19 05:44:33 2007 From: deveshbissa at rediffmail.com (devesh bissa) Date: 19 Oct 2007 05:44:33 -0000 Subject: [SIPForum-discussion] Re:CODEC Message-ID: <20071019054433.11267.qmail@f5mail16.rediffmail.com> Hi,SIP is purely signalling protocol,and we use codec for media protocol. Is it correct?Please verify me.Thank you,Devesh Devesh Bissa -------------- next part -------------- An HTML attachment was scrubbed... URL: From jeancosta at gmail.com Fri Oct 19 11:39:09 2007 From: jeancosta at gmail.com (Jean Rodrigo) Date: Fri, 19 Oct 2007 08:39:09 -0300 Subject: [SIPForum-discussion] Instructions about VoiceRD - Asterisk Message-ID: Hi everybody! Does anyone have any experience with the software VoiceRD? I'm trying to install this and create an enviroment to take the authentication process (register) out of asterisk and do it at a LDAP directory. I'll appreciate any kind of help! Thank you! Jean Costa -------------- next part -------------- An HTML attachment was scrubbed... URL: From inako at abcom.al Fri Oct 19 12:07:23 2007 From: inako at abcom.al (Ilir Nako) Date: Fri, 19 Oct 2007 14:07:23 +0200 Subject: [SIPForum-discussion] How to implement callshop References: Message-ID: <006001c81248$9bcd56d0$0a464e50@IlirNako> Hi all I want to implement callshop .I'm serching in internet but have not decided how to implemnt it. I need some help eg: the billing software the voip gateway a server when i connect the accounts of my customers and to use this soft like proxy to use my account with one voip service provider. ??? Best regards Ilir -------------- next part -------------- An HTML attachment was scrubbed... URL: From Dan.Broussard at Level3.com Fri Oct 19 12:48:54 2007 From: Dan.Broussard at Level3.com (Broussard, Dan) Date: Fri, 19 Oct 2007 06:48:54 -0600 Subject: [SIPForum-discussion] How to implement callshop Message-ID: <5EDAA8E0E2355946B2296BFC33307F851A8F5C@idc1exc0006.corp.global.level3.com> Voip provider http://www.reignmaker.net ----- Original Message ----- From: discussion-bounces at sipforum.org To: discussion at sipforum.org Sent: Fri Oct 19 06:07:23 2007 Subject: [SIPForum-discussion] How to implement callshop Hi all I want to implement callshop .I'm serching in internet but have not decided how to implemnt it. I need some help eg: the billing software the voip gateway a server when i connect the accounts of my customers and to use this soft like proxy to use my account with one voip service provider. ??? Best regards Ilir From victor.pascual.avila at gmail.com Fri Oct 19 15:03:01 2007 From: victor.pascual.avila at gmail.com (=?ISO-8859-1?Q?Victor_Pascual_=C1vila?=) Date: Fri, 19 Oct 2007 17:03:01 +0200 Subject: [SIPForum-discussion] How to implement callshop In-Reply-To: <5EDAA8E0E2355946B2296BFC33307F851A8F5C@idc1exc0006.corp.global.level3.com> References: <5EDAA8E0E2355946B2296BFC33307F851A8F5C@idc1exc0006.corp.global.level3.com> Message-ID: <618e24240710190803k2226cbf5l19ba33055b294d5f@mail.gmail.com> I suggest you to contract a TelephonyServiceProvider (eg. www.voztele.com) and outsource all the VoIP platform if you are not an expert in that. Due most calls will be PSTN routed, it has no sense in deploying your own SIP platform. You could ask you TSP to provide you CPE equipment (for example Linksys PAP2) with reverse polarity enabled. Then, you can connect a billing device between the PAP2 and the analogic phone. This could be the scenario: phone---billing equipment---LinksysPAP2---router---internet---TelephonyOverIP Service Prov I hope it was useful, Kind regards, Victor Pascual On 19/10/2007, Broussard, Dan wrote: > Voip provider http://www.reignmaker.net > > > > > ----- Original Message ----- > From: discussion-bounces at sipforum.org > To: discussion at sipforum.org > Sent: Fri Oct 19 06:07:23 2007 > Subject: [SIPForum-discussion] How to implement callshop > > Hi all > > I want to implement callshop .I'm serching in internet but have not decided how to implemnt it. > I need some help eg: > > the billing software > the voip gateway > a server when i connect the accounts of my customers and to use this soft like proxy to use my account with one > voip service provider. > ??? > > Best regards Ilir > > _______________________________________________ > This is the SIP Forum discussion mailing list > TO UNSUBSCRIBE, or edit your delivery options, please visit http://sipforum.org/mailman/listinfo/discussion > Post to the list at discussion at sipforum.org > From jacksont at binarytelecom.com Fri Oct 19 15:32:00 2007 From: jacksont at binarytelecom.com (Tim Jackson) Date: Fri, 19 Oct 2007 08:32:00 -0700 Subject: [SIPForum-discussion] Instructions about VoiceRD - Asterisk In-Reply-To: Message-ID: <0MKpCa-1IitpO20C0-0004oj@mrelay.perfora.net> www.infoarch.com These guys are Asterisk consultants who will consult over the phone, and have reasonable rates. Tim Jackson Binary Telecom, Inc. 3300 NW 185th #197 Portland, OR 97229 www.binarytelecom.com jacksont at binarytelecom.com 800.594.3670 (toll-free) 503.268.0287 (Direct) 503.564.4534 (fax) _____ From: discussion-bounces at sipforum.org [mailto:discussion-bounces at sipforum.org] On Behalf Of Jean Rodrigo Sent: Friday, October 19, 2007 4:39 AM To: discussion at sipforum.org Subject: [SIPForum-discussion] Instructions about VoiceRD - Asterisk Hi everybody! Does anyone have any experience with the software VoiceRD? I'm trying to install this and create an enviroment to take the authentication process (register) out of asterisk and do it at a LDAP directory. I'll appreciate any kind of help! Thank you! Jean Costa -------------- next part -------------- An HTML attachment was scrubbed... URL: From stephen.mcvarnock at aepona.com Fri Oct 19 15:34:38 2007 From: stephen.mcvarnock at aepona.com (Stephen McVarnock) Date: Fri, 19 Oct 2007 16:34:38 +0100 Subject: [SIPForum-discussion] Route header and 'orig' parameter Message-ID: <4718CE8E.2020502@aepona.com> Hi folks, Trying to find details on the use of the 'orig' parameter in the Route header for IMS (AS -> S-CSCF). TS 24.229, section 5.7.3 covers it at a high level but I'm looking for something with more detail on the use of the orig/term parameter. Thanks in advance, Steve. From juan.freitas at novabase.pt Fri Oct 19 16:16:37 2007 From: juan.freitas at novabase.pt (Juan Freitas) Date: Fri, 19 Oct 2007 17:16:37 +0100 Subject: [SIPForum-discussion] SIP/SIMPLE client Message-ID: Hi everyone. I need a SIP/SIMPLE client source code to perform some tests. I just want a very simple client that can see other contacts status. Anyone has one that can share? I know that sip-communicator and do this but I need a simpler client to test a combination of changes to the code rapidly. Thanks, Juan Freitas Analyst ........................................................................ ..................................... Novabase Av. Eng. Duarte Pacheco, 15F . 1099-078 Lisboa - Portugal Tel. (+351) 213 836 300 . Fax (+351) 213 836 301 . mailto:juan.freitas at novabase.pt www.novabase.pt -------------- next part -------------- An HTML attachment was scrubbed... URL: From jonathanpwagner at gmail.com Fri Oct 19 16:42:58 2007 From: jonathanpwagner at gmail.com (Jonathan Wagner) Date: Fri, 19 Oct 2007 12:42:58 -0400 Subject: [SIPForum-discussion] Route header and 'orig' parameter In-Reply-To: <4718CE8E.2020502@aepona.com> References: <4718CE8E.2020502@aepona.com> Message-ID: The orig and term headers indicate to the AS what features should be applied (originating or terminating). So the S-CSCF sends the call to the AS that will apply orginating services and signifies that with 'orig' (i.e. Calling Line ID, call restrictions, etc.), the call is sent back down to the s-CSCF that will continue running the originating service triggers, once complete, the CSCF will begin to run terminating triggers and if necessary send the call to the AS that will apply terminating services and signify that leg as 'term'. On 10/19/07, Stephen McVarnock wrote: > > Hi folks, > > Trying to find details on the use of the 'orig' parameter in the Route > header > for IMS (AS -> S-CSCF). > > TS 24.229, section 5.7.3 covers it at a high level but I'm looking for > something > with more detail on the use of the orig/term parameter. > > Thanks in advance, > Steve. > _______________________________________________ > This is the SIP Forum discussion mailing list > TO UNSUBSCRIBE, or edit your delivery options, please visit > http://sipforum.org/mailman/listinfo/discussion > Post to the list at discussion at sipforum.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From govindraj_h at yahoo.co.in Mon Oct 22 01:04:25 2007 From: govindraj_h at yahoo.co.in (Govindraj.B.H @ Gkk) Date: Mon, 22 Oct 2007 06:34:25 +0530 (IST) Subject: [SIPForum-discussion] CODEC Message-ID: <734078.33388.qm@web8414.mail.in.yahoo.com> Yes you are right. SIP is a signaling protocol and codecs are negotiated during the session establishment. These codecs do coding/decoding of RTP packets. All, if I am wrong plz correct me. Regards Govindraj B H ----- Original Message ---- From: devesh bissa To: discussion at sipforum.org Sent: Friday, 19 October, 2007 1:44:33 AM Subject: [SIPForum-discussion] Re:CODEC Hi, SIP is purely signalling protocol,and we use codec for media protocol. Is it correct? Please verify me. Thank you, Devesh Devesh Bissa -----Inline Attachment Follows----- _______________________________________________ This is the SIP Forum discussion mailing list TO UNSUBSCRIBE, or edit your delivery options, please visit http://sipforum.org/mailman/listinfo/discussion Post to the list at discussion at sipforum.org Bring your gang together - do your thing. Go to http://in.promos.yahoo.com/groups -------------- next part -------------- An HTML attachment was scrubbed... URL: From govindraj_h at yahoo.co.in Mon Oct 22 01:10:30 2007 From: govindraj_h at yahoo.co.in (Govindraj.B.H @ Gkk) Date: Mon, 22 Oct 2007 06:40:30 +0530 (IST) Subject: [SIPForum-discussion] CODEC Message-ID: <971052.46899.qm@web8403.mail.in.yahoo.com> G711...........audio codec. G729 ( G729a, G729b, G729ab ) -----audio codecs and T.38........Fax codec. ----- Original Message ---- From: fan yongjie To: ?The Passenger? Cc: discussion at sipforum.org Sent: Thursday, 18 October, 2007 11:03:19 PM Subject: Re: [SIPForum-discussion] CODEC SIP does not support codecs directly. If you want to support some codecs, you can try to get some free project to implement it. for example, pjmedia the normal codecs include, G.711u/a, G.729, etc. 2007/10/19, ?The Passenger? : Hi every body is any one can tell me what is the codec which are already supported by SIP ? Audio and Video Codec also, i know its alot but please if you can memorize some of them thanks alot ((mOh AmAd fAiZ )) __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com _______________________________________________ This is the SIP Forum discussion mailing list TO UNSUBSCRIBE, or edit your delivery options, please visit http://sipforum.org/mailman/listinfo/discussion Post to the list at discussion at sipforum.org -----Inline Attachment Follows----- _______________________________________________ This is the SIP Forum discussion mailing list TO UNSUBSCRIBE, or edit your delivery options, please visit http://sipforum.org/mailman/listinfo/discussion Post to the list at discussion at sipforum.org Bring your gang together - do your thing. Go to http://in.promos.yahoo.com/groups -------------- next part -------------- An HTML attachment was scrubbed... URL: From GBeith at empirix.com Mon Oct 22 02:09:03 2007 From: GBeith at empirix.com (Beith, Gordon) Date: Sun, 21 Oct 2007 22:09:03 -0400 Subject: [SIPForum-discussion] CODEC In-Reply-To: <971052.46899.qm@web8403.mail.in.yahoo.com> References: <971052.46899.qm@web8403.mail.in.yahoo.com> Message-ID: There are a bunch more than this....just look at all the IETF codec specs that describe the SDP formats/usages. Many of them are wireless codecs. ________________________________ From: discussion-bounces at sipforum.org [mailto:discussion-bounces at sipforum.org] On Behalf Of Govindraj.B.H @ Gkk Sent: Sunday, October 21, 2007 9:11 PM To: fan yongjie; ?The Passenger? Cc: discussion at sipforum.org Subject: Re: [SIPForum-discussion] CODEC G711...........audio codec. G729 ( G729a, G729b, G729ab ) -----audio codecs and T.38.........Fax codec. ----- Original Message ---- From: fan yongjie To: ?The Passenger? Cc: discussion at sipforum.org Sent: Thursday, 18 October, 2007 11:03:19 PM Subject: Re: [SIPForum-discussion] CODEC SIP does not support codecs directly. If you want to support some codecs, you can try to get some free project to implement it. for example, pjmedia the normal codecs include, G.711u/a, G.729, etc. 2007/10/19, ?The Passenger? : Hi every body is any one can tell me what is the codec which are already supported by SIP ? Audio and Video Codec also, i know its alot but please if you can memorize some of them thanks alot ((mOh AmAd fAiZ )) __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com _______________________________________________ This is the SIP Forum discussion mailing list TO UNSUBSCRIBE, or edit your delivery options, please visit http://sipforum.org/mailman/listinfo/discussion Post to the list at discussion at sipforum.org -----Inline Attachment Follows----- _______________________________________________ This is the SIP Forum discussion mailing list TO UNSUBSCRIBE, or edit your delivery options, please visit http://sipforum.org/mailman/listinfo/discussion Post to the list at discussion at sipforum.org ________________________________ Bring your gang together - do your thing. Start your group. -------------- next part -------------- An HTML attachment was scrubbed... URL: From vendors at tpsoft.com Mon Oct 22 02:24:01 2007 From: vendors at tpsoft.com (Barry Demchak) Date: Sun, 21 Oct 2007 19:24:01 -0700 Subject: [SIPForum-discussion] Calls vs sessions vs dialogs vs transactions Message-ID: <7.0.1.0.2.20071021191710.034309b0@tpsoft.com> Hi, all -- Sorry for such an elementary question. I'm trying to model some aspects of SIP, and I'm not quite sure how calls, sessions, and dialogs relate. (Dialogs are easy ... they're the context for transactions ... and can support a number of them over time.) I have definitely read RFP3261, but it's not quite helpful here. I have also visited the (excellent) tech-info site. Here's what I'm having trouble with: Is a call the same as a session?? Apparently a session can have numerous dialogs ... when can that happen, and what does it mean? Clearly, setting up a dialog has a formal process. Is there a process for constructing a session or a call ... separate from setting up a dialog?? Thanks. From tuanna at avagroup.vn Tue Oct 23 00:37:37 2007 From: tuanna at avagroup.vn (Nguyen Anh Tuan) Date: Tue, 23 Oct 2007 07:37:37 +0700 Subject: [SIPForum-discussion] (no subject) Message-ID: <7BEC1A7B-F1A4-4D4C-A522-59784F164597@avagroup.vn> I want to use a mobicents sip server (https://mobicents.dev.java.net/ base jboss), however the billing module not found. If who used the mobicent, please help me. Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: From david.calhoun at cinbell.com Tue Oct 23 02:03:12 2007 From: david.calhoun at cinbell.com (david.calhoun at cinbell.com) Date: Mon, 22 Oct 2007 22:03:12 -0400 Subject: [SIPForum-discussion] UC500 Message-ID: Hi, I'm trying to provide SIP trunking to the UC500, which Broadsoft does not have a PCG for yet (due late this year). Can anyone provide what the configs should look like to have SIP trunks working on the UC500? I can't get the UC500 to even register at this point. Thank you, Dave Calhoun Specialist - Integrated Planner Cincinnati Bell Telephone 209 W7th Street Mail Stop 121-425 Cincinnati, OH 45201 Office: 513.565.2441 Mobile: 513.477.0495 E-mail: david.calhoun at cinbell.com The information transmitted is intended only for the person or entity to which it is addressed and may contain confidential and/or privileged material. Any review, retransmission, dissemination or other use of, or taking of any action in reliance upon, this information by persons or entities other than the intended recipient is prohibited. If you receive this in error, please contact the sender and destroy any copies of this document. From voiptraining at gmail.com Tue Oct 23 04:49:16 2007 From: voiptraining at gmail.com (Kumar DN) Date: Tue, 23 Oct 2007 10:19:16 +0530 Subject: [SIPForum-discussion] Announcing VoIP/SIP training program in Hyderabad, INDIA Message-ID: <957f17eb0710222149t4c2ae042i9e35d2d32e1b5146@mail.gmail.com> Hello SIP Lovers, *Announcing VoIP/SIP Conceptual Course:* I hereby announce the Telecom/VoIP/SIP conceptual course for the job seekers. The ideal candidates for the course should belong to Electronics and Communication Engineering, Computer Science Engineering, and Information Technology Engineering. Please find the attached word document, "VoIP-SIP course content". The classroom training will take place at the following address: Ocean Technologies, Flat No. 501, Above City Financial, Opposite side of S.R. Nagar Police Station, Hyderabad ? 500 038, INDIA Mobile : +91-94400-30971 E-mail : voiptraining at gmail.com With warm regards, Kumar. -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: VoIP-SIP course.doc Type: application/msword Size: 42496 bytes Desc: not available URL: From asha.g.pillai at gmail.com Tue Oct 23 11:54:56 2007 From: asha.g.pillai at gmail.com (Asha G) Date: 23 Oct 2007 04:54:56 -0700 Subject: [SIPForum-discussion] Friendship Request on Shelfari Message-ID: <200710231155.l9NBt6xm029826@sipforum.org> An HTML attachment was scrubbed... URL: From priyank_mvit at rediffmail.com Tue Oct 23 13:18:13 2007 From: priyank_mvit at rediffmail.com (priyank gupta) Date: 23 Oct 2007 13:18:13 -0000 Subject: [SIPForum-discussion] Regarding Darwin Streaming Server Message-ID: <20071023131813.30450.qmail@f5mail-237-207.rediffmail.com> ? Hi, I have darwin streaming server installed in one of the system. Can any body tell me how to stream the data using Darwin Server...? Which codec Darwin Server will support...? Can we use any file for streaming in Darwin Server...? I read any article on net regarding including hints to play any file in darwin server....? can any body explain me what is this hint and how it helps in streaming a file...? and one more thing what is the difference between Darwin Streaming Server and Quick Time Streaming Server.....? Plz refer any document regarding all this....? thanks Regards Priyank Gupta RnD Department LnT Infotech Limited Bangalore -------------- next part -------------- An HTML attachment was scrubbed... URL: From rjsparks at nostrum.com Tue Oct 23 14:14:48 2007 From: rjsparks at nostrum.com (Robert Sparks) Date: Tue, 23 Oct 2007 09:14:48 -0500 Subject: [SIPForum-discussion] SIPit 21 registration closes in less than a week Message-ID: <83A5BDBC-AB50-497A-877F-135F7E8C5AD0@nostrum.com> The registration deadline for SIPit 21 is Oct 29, 6 days from now. If you have not already registered, but plan to attend, please register now. (If you can't register until later in the week, drop me a note - I need to get some information for the event setup from you early). RjS From ashoke.k.ghosh at gmail.com Tue Oct 23 14:27:11 2007 From: ashoke.k.ghosh at gmail.com (Ashoke Kumar Ghosh) Date: Tue, 23 Oct 2007 19:57:11 +0530 Subject: [SIPForum-discussion] SIP stack architecture Message-ID: <004d01c81580$d152f090$da31e0dc@ASHOKE> Hi, Can anybody provide some document on sip stack architecture or refer to some link. Best Regards.. ******************************************* Ashoke Kumar Ghosh Mobile : 919324279664 Email: ashoke.k.ghosh at gmail.com ******************************************* -------------- next part -------------- An HTML attachment was scrubbed... URL: From tuanna at avagroup.vn Tue Oct 23 17:09:54 2007 From: tuanna at avagroup.vn (Nguyen Anh Tuan) Date: Wed, 24 Oct 2007 00:09:54 +0700 Subject: [SIPForum-discussion] SIP stack architecture In-Reply-To: <004d01c81580$d152f090$da31e0dc@ASHOKE> Message-ID: You can search on winkipedia! _____ From: discussion-bounces at sipforum.org [mailto:discussion-bounces at sipforum.org] On Behalf Of Ashoke Kumar Ghosh Sent: Tuesday, October 23, 2007 9:27 PM To: discussion at sipforum.org Subject: [SIPForum-discussion] SIP stack architecture Hi, Can anybody provide some document on sip stack architecture or refer to some link. Best Regards.. ******************************************* Ashoke Kumar Ghosh Mobile : 919324279664 Email: ashoke.k.ghosh at gmail.com ******************************************* -------------- next part -------------- An HTML attachment was scrubbed... URL: From victor.pascual.avila at gmail.com Tue Oct 23 18:17:03 2007 From: victor.pascual.avila at gmail.com (=?ISO-8859-1?Q?Victor_Pascual_=C1vila?=) Date: Tue, 23 Oct 2007 20:17:03 +0200 Subject: [SIPForum-discussion] Regarding Darwin Streaming Server In-Reply-To: <20071023131813.30450.qmail@f5mail-237-207.rediffmail.com> References: <20071023131813.30450.qmail@f5mail-237-207.rediffmail.com> Message-ID: <618e24240710231117v5f70fe59q8030d63958d03f33@mail.gmail.com> Sorry, could you remember us if Darwin supports only RTSP or supports SIP as well? http://developer.apple.com/opensource/server/streaming/index.html Thanks, Victor On 23 Oct 2007 13:18:13 -0000, priyank gupta wrote: > > > > Hi, > > I have darwin streaming server installed in one of the system. > Can any body tell me how to stream the data using Darwin Server...? > Which codec Darwin Server will support...? > > Can we use any file for streaming in Darwin Server...? > I read any article on net regarding including hints to play any file in > darwin server....? > > can any body explain me what is this hint and how it helps in streaming a > file...? > > and one more thing > what is the difference between Darwin Streaming Server and Quick Time > Streaming Server.....? > > Plz refer any document regarding all this....? > > thanks > Regards > > Priyank Gupta > RnD Department > LnT Infotech Limited > Bangalore > > > > _______________________________________________ > This is the SIP Forum discussion mailing list > TO UNSUBSCRIBE, or edit your delivery options, please visit > http://sipforum.org/mailman/listinfo/discussion > Post to the list at discussion at sipforum.org > > From vendors at tpsoft.com Tue Oct 23 20:07:50 2007 From: vendors at tpsoft.com (Barry Demchak) Date: Tue, 23 Oct 2007 13:07:50 -0700 Subject: [SIPForum-discussion] Calls vs sessions vs dialogs vs transactions Message-ID: <7.0.1.0.2.20071023130738.03456928@tpsoft.com> Thanks, Deepanshu ... the draft-ietf-sipping-dialogusage-04 was very informative. From RFC3261 and from the example in http://www.tech-invite.com/Ti-sip-service-11.html, this is what it looks like to me: (Anyone: feel free to jump in) A dialog (1 usage) is used to set up and maintain a session. A session is an exchange of data, including voice or video. The point of the dialog is to set up the session and then stay out of the way. So, the real feed is the session, and the dialog amounts to out-of-band signalling. A dialog can contain multiple transactions, staged serially. A transaction can affect the dialog and/or the session. There isn't a formal definition of the term "call", though a call-id is part of a dialog identifier. The sense of "call" I get from http://www.tech-invite.com/Ti-sip-service-11.html is that it can contain multiple dialogs (as in the case of a dialog between Alice and Bob, and another between Bob and Carol). As for the relationship between dialogs and usages, a dialog can contain one or more usages, and when the last usage closes, the dialog closes, too. Comments?? Deepanshu?? Anyone else?? Thanks. At 02:29 AM 10/23/2007, Deepanshu wrote: >calls and session are same. > >session can have numerous dialogs in the following case: > >A -----INVITE---------->B >A<-----200OK------------B >-------session established--------- >-------INVITE Dialog established -------- >A---------REFER--------->B >A<---------202 Accepted--------B >--------REFER Dialog established with in current session---------- >A <------------BYE------------B >--------------session/refer dialog/invite dialog ends----------------- > >refer to draft-ietf-sipping-dialogusage-04 for more > >BR >Deepanshu Gautam >Huawei Technologies Co. Ltd. > > > >----- Original Message ----- >From: "Barry Demchak" >To: >Sent: Monday, October 22, 2007 10:24 AM >Subject: [SIPForum-discussion] Calls vs sessions vs dialogs vs transactions > > > > Hi, all -- > > > > Sorry for such an elementary question. I'm trying to model some > > aspects of SIP, and I'm not quite sure how calls, sessions, and > > dialogs relate. (Dialogs are easy ... they're the context for > > transactions ... and can support a number of them over time.) > > > > I have definitely read RFP3261, but it's not quite helpful here. I > > have also visited the (excellent) tech-info site. > > > > Here's what I'm having trouble with: > > > > Is a call the same as a session?? > > > > Apparently a session can have numerous dialogs ... when can that > > happen, and what does it mean? > > > > Clearly, setting up a dialog has a formal process. Is there a process > > for constructing a session or a call ... separate from setting up a >dialog?? > > > > Thanks. > > > > > > > > _______________________________________________ > > This is the SIP Forum discussion mailing list > > TO UNSUBSCRIBE, or edit your delivery options, please visit >http://sipforum.org/mailman/listinfo/discussion > > Post to the list at discussion at sipforum.org From deepanshu at huawei.com Wed Oct 24 01:58:32 2007 From: deepanshu at huawei.com (Deepanshu) Date: Wed, 24 Oct 2007 09:58:32 +0800 Subject: [SIPForum-discussion] Calls vs sessions vs dialogs vs transactions References: <7.0.1.0.2.20071021191710.034309b0@tpsoft.com> <004d01c81557$42aa3c50$9e78a40a@china.huawei.com> <7.0.1.0.2.20071023124254.03556f30@tpsoft.com> Message-ID: <004f01c815e1$61b90740$9e78a40a@china.huawei.com> inine line staring in [DG] ----- Original Message ----- From: "Barry Demchak" To: "Deepanshu" Sent: Wednesday, October 24, 2007 4:06 AM Subject: Re: [SIPForum-discussion] Calls vs sessions vs dialogs vs transactions > Thanks, Deepanshu ... the draft-ietf-sipping-dialogusage-04 was very > informative. > > From RFC3261 and from the example in > http://www.tech-invite.com/Ti-sip-service-11.html, this is what it > looks like to me: > > (Anyone: feel free to jump in) > > A dialog (1 usage) is used to set up and maintain a session. > > A session is an exchange of data, including voice or video. The point > of the dialog is to set up the session and then stay out of the way. > So, the real feed is the session, and the dialog amounts to > out-of-band signalling. > > A dialog can contain multiple transactions, staged serially. A > transaction can affect the dialog and/or the session. > > There isn't a formal definition of the term "call", though a call-id > is part of a dialog identifier. The sense of "call" I get from > http://www.tech-invite.com/Ti-sip-service-11.html is that it can > contain multiple dialogs (as in the case of a dialog between Alice > and Bob, and another between Bob and Carol). [DG] as you said there 'call' is not a formal term in SIP, so anyone can put that in his/her own way. According to me 'call' is neither a session nor a dialog rather it is just a user action. > > As for the relationship between dialogs and usages, a dialog can > contain one or more usages, and when the last usage closes, the > dialog closes, too. [DG] that true. I would like to add one point. This type of behaviour is being criticized because of its complex nature. In draft-ietf-sipping-dialogusage-04 it is said to aviod using dialog with multiple usage. The possible solution is to use Target-Dialog header filed (RFC4538) > > Comments?? Deepanshu?? Anyone else?? > > Thanks. > > At 02:29 AM 10/23/2007, Deepanshu wrote: > >calls and session are same. > > > >session can have numerous dialogs in the following case: > > > >A -----INVITE---------->B > >A<-----200OK------------B > >-------session established--------- > >-------INVITE Dialog established -------- > >A---------REFER--------->B > >A<---------202 Accepted--------B > >--------REFER Dialog established with in current session---------- > >A <------------BYE------------B > >--------------session/refer dialog/invite dialog ends----------------- > > > >refer to draft-ietf-sipping-dialogusage-04 for more > > > >BR > >Deepanshu Gautam > >Huawei Technologies Co. Ltd. > > > > > > > >----- Original Message ----- > >From: "Barry Demchak" > >To: > >Sent: Monday, October 22, 2007 10:24 AM > >Subject: [SIPForum-discussion] Calls vs sessions vs dialogs vs transactions > > > > > > > Hi, all -- > > > > > > Sorry for such an elementary question. I'm trying to model some > > > aspects of SIP, and I'm not quite sure how calls, sessions, and > > > dialogs relate. (Dialogs are easy ... they're the context for > > > transactions ... and can support a number of them over time.) > > > > > > I have definitely read RFP3261, but it's not quite helpful here. I > > > have also visited the (excellent) tech-info site. > > > > > > Here's what I'm having trouble with: > > > > > > Is a call the same as a session?? > > > > > > Apparently a session can have numerous dialogs ... when can that > > > happen, and what does it mean? > > > > > > Clearly, setting up a dialog has a formal process. Is there a process > > > for constructing a session or a call ... separate from setting up a > >dialog?? > > > > > > Thanks. > > > > > > > > > > > > _______________________________________________ > > > This is the SIP Forum discussion mailing list > > > TO UNSUBSCRIBE, or edit your delivery options, please visit > >http://sipforum.org/mailman/listinfo/discussion > > > Post to the list at discussion at sipforum.org > From jeancosta at gmail.com Wed Oct 24 14:01:34 2007 From: jeancosta at gmail.com (Jean Rodrigo) Date: Wed, 24 Oct 2007 11:01:34 -0300 Subject: [SIPForum-discussion] VoiceRD and Edirectory 8.8.1 Instalation Message-ID: Hi everybody! I'm trying to develop an environment where I can use Asterisk integrated to an Ldap directory. I found the VoiceRD software that provides it but I'm not having success in the instalation. Does anyone have any tutorial that could help me to install it? At this moment I'm trying to install the Novell Edirectory 8.8.1 but it gives the message below: [root at localhost setup]# ./nds-install -c server -c admutils -u ./nds-install: line 669: [: too many arguments ./nds-install: line 691: [: too many arguments ./nds-install: line 691: [: too many arguments %%% There are no components available to install. %%% Some packages required for the components may be missing in I'll appreciate any kind of help! Thank you!! Jean Costa. -------------- next part -------------- An HTML attachment was scrubbed... URL: