[SIPForum-discussion] # converted to %2 in SIP invite

Tue Aug 12 12:56:23 UTC 2014

Hi

This is absolutely correct. Actually the 23 (Hex) is the ASCII code for character #, and as you see in the INVITE message there is the %23 in front of the dialed  digits, which is the # using URL encoding.

-- 
Dimitris Bourdoukis 

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Today's Topics:

   1. Re: # converted to %2 in SIP invite (qcorba)
   2. Difference between Queued and waiting (Tapender Kumar)
   3. Re: Doubt regarding Via. (tester voip)
   4. Re: Sipp with stateless proxy (tester voip)

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Message: 1
Date: Wed, 6 Aug 2014 13:56:35 +0800
From: qcorba <qcorba at gmail.com>
Subject: Re: [SIPForum-discussion] # converted to %2 in SIP invite
To: Ramachandra moorthy <ramcm.irtt at gmail.com>
Cc: "discussion at sipforum.org" <discussion at sipforum.org>
Message-ID:
	<CANjJQeYA=1dkA5xcxzkaUt+J2ed0nKYwcm0O=r8VQNhffuYFLw at mail.gmail.com>
Content-Type: text/plain; charset="utf-8"

Reserved characters like # are translated to % encoding. Please see:
http://en.m.wikipedia.org/wiki/Percent-encoding

Ramachandra moorthy <ramcm.irtt at gmail.com> ? 2014?7?17???? ???

> Dear All,
>
> Kindly clarify my another query pls..
>
> SIP subscriber dialing" #" sympol...but it is transfer like "%2" in 
> SIP invite message..Kindly clarify whether this is right or wrong.
>
> Below is the invite message
>
> Sequence NO.:1
> INVITE sip:2%232008 at 10.191.59.22:5060;user=phone SIP/2.0
> Via: SIP/2.0/UDP 10.190.32.102:5060;branch=z9hG4bK7d9501b5.0
> To: "2%232008"<sip:2%232008 at 10.190.32.102
> <javascript:_e(%7B%7D,'cvml','sip:2%25232008 at 10.190.32.102');>>
> From: "4023325578"<sip:4023325578 at 10.190.32.102
> <javascript:_e(%7B%7D,'cvml','sip:4023325578 at 10.190.32.102');>
> >;tag=0abe2066-000011b200006f84
> Call-ID: 0000052c000049e9-0047-0791 at 10.190.32.102
> <javascript:_e(%7B%7D,'cvml','0000052c000049e9-0047-0791 at 10.190.32.102
> ');>
> CSeq: 6673 INVITE
> Contact: <sip:4023325578 at 10.190.32.102:5060>
> Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,REFER,NOTIFY,PRACK,UPDATE
> P-Asserted-Identity: <sip:4023325578 at 10.190.32.102 
> <javascript:_e(%7B%7D,'cvml','sip:4023325578 at 10.190.32.102');>>
> Max-Forwards: 9
> Record-Route: <sip:10.190.32.102:5060;lr>
> Supported: 100rel
> User-Agent: ZTE Softswitch/1.0.0
> P-Charging-Vector: icid-value=Hyder-20140717125720-0001eb81
> Content-Type: multipart/mixed;boundary=zte-unique-boundary-06
> Content-Length: 515
>
> --zte-unique-boundary-06
> Content-Type: application/SDP
>
> v=0
> o=ZTE 4 27240 IN IP4 10.190.32.102
> s=phone-call
> c=IN IP4 10.190.32.22
> t=0 0
> m=audio 19658 RTP/AVP 8 18 97
> a=rtpmap:97 telephone-event/8000
> a=fmtp:97 0-11
> a=ptime:20
>
> --zte-unique-boundary-06
> Content-Type: application/ISUP; base=nxv3; version=itu-t92+
> Content-Disposition: signal; handling=optional
>
>
> /*----------start isup message data----------
> 0000: 01 00 60 00 0a 03 02 07 05 01 90 c2 02 80 3d 01
> 0010: 1e 0a 07 03 13 04 32 23 55 87 c6 07 00 00 01 59
> 0020: 00 00 00 31 02 00 00 78 18 e2 82 c0 01 a1 12 02
> 0030: 01 00 02 01 01 30 0a 83 01 00 87 01 10 a9 02 84
> 0040: 00 78 21 81 80 c0 02 81 1c 1a 91 aa 06 80 01 00
> 0050: 82 01 00 8b 01 00 a1 0c 02 02 40 67 06 04 2b 0c
> 0060: 09 00 84 00 39 08 31 c0 c6 d0 3d c0 78 c4 00 IAM----Initial 
> address message
>
> Mandatory parameter:
> Nature of connection indicators:
> Satellite indicator:0 satellite circuits in the connection Continuity 
> check indicator:Continuity check not required Echo control device 
> indicator:Not Include
>
> Forward call indicators:
> International/national call indicator:national call End-to-end method 
> indicator: Not available Interworking indicator: Interworking not 
> encountered End-to-end information indicator: Not available ISDN user 
> part indicator: ISDN user part used all the way ISDN user part 
> preference indicator: ISDN user part not required all the way ISDN 
> access indicator: Originating access non-ISDN SCCP method indication: 
> No indication Collect Call:0
>
> Calling party category:Ordinary subscriber Transmission medium 
> requirement:3.1 kHz audio
>
> Called party number:
> Nature of address: subscriber number
> Internal network number indicator (INN):   *routing to internal network
> number not allowed
> Numbering plan indicator: ISDN numbering plan Address signal: 2#2008
>
> Optional parameter:
> Hop counter: 30
> Calling party number:
> Nature of address: National (significant) number
> Calling party number incomplete indicator (NI):   *Complete
> Numbering plan indicator: ISDN numbering plan Address presentation 
> restricted indicator: Presentation allowed Screening indicators: 
> Network provided Address signal: 4023325578 Propagation delay counter:  
> 0 ms Parameter compatibility Information:
> Parameter#1
> Propagation delay counter:
> Transit at intermediate exchange indicator:  Transit interpretation 
> Release call indicator:  Not release call Send notification:  Not Send 
> notification Discard message indicator:  Not Discard message Discard 
> parameter indicator:do not discard parameter (pass on) Pass on not 
> possible indicator:discard parameter
>
> *----------end isup message data----------*/
>
> --zte-unique-boundary-06--
>
>
>
>
>
> *With Regards M. RamachandramoorthyNGN-BSNL+91- 9600070877*
>
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Message: 2
Date: Thu, 31 Jul 2014 13:00:29 +0530
From: Tapender Kumar <tapender5002 at gmail.com>
Subject: [SIPForum-discussion] Difference between Queued and waiting
To: "discussion at sipforum.org" <discussion at sipforum.org>
Message-ID:
	<CAH7Fa8DgFsVoa4-uoXN=LjZyP1usyVTi5JcF1Tncb0jfEK5thw at mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"

Dear All,

Can any one explain difference about Queued and waiting.

--
With Regards,
Tapender Kumar

*
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Message: 3
Date: Sat, 9 Aug 2014 15:43:09 +0530
From: tester voip <tester.voip1 at gmail.com>
Subject: Re: [SIPForum-discussion] Doubt regarding Via.
To: Tamoghna Bhaduri <tamoghnabhaduri at gmail.com>
Cc: "discussion at sipforum.org" <discussion at sipforum.org>
Message-ID:
	<CALndBVaKBBLcOXa_SEQxrTrYbcnpA5KU6CEFLsFcCOAQjfcdYA at mail.gmail.com>
Content-Type: text/plain; charset="utf-8"

yes.if the last node is a backtoback,sbc

On Tue, Jun 17, 2014 at 3:54 AM, Tamoghna Bhaduri <tamoghnabhaduri at gmail.com
> wrote:

> Hi All,
> I have the below doubt regarding Via header in case of INVITE received at
> MT side.
> As per my basic knowledge at MT side we will see multiple Via,as each node
> add there own Via
> to the INVITE request.
>
> But is it possible in any case that at MT side only 1 Via will be
> there(the last adjacent Node's Via).
>
>
>
>
>
>
>
>
>
> _______________________________________________
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> TO UNSUBSCRIBE, or edit your delivery options, please visit
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>
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Message: 4
Date: Sat, 9 Aug 2014 15:54:48 +0530
From: tester voip <tester.voip1 at gmail.com>
Subject: Re: [SIPForum-discussion] Sipp with stateless proxy
To: Aditya Prakash <adipra90 at gmail.com>
Cc: "discussion at sipforum.org" <discussion at sipforum.org>, "Kapoor,
	Geetika" <geetika.kapoor at hp.com>
Message-ID:
	<CALndBVaqUSCfXHLt3DY=8L+qcxmrfaeynzQ_gG=WY7B-2bbhzQ at mail.gmail.com>
Content-Type: text/plain; charset="utf-8"

yes there can b stateles proxy unles it uses recor-route.

there can be atleast a dialog per sesion.but can be more as with the cases
of subscribe-notify style dialogs withn the ses

On Fri, Jun 27, 2014 at 11:43 AM, Aditya Prakash <adipra90 at gmail.com> wrote:

> hi all
> can any one plz explain me what does it mean "maximum dialog in as
> session"..
> according to my knowledge wen UA sends INVITE request and 200 OK comes
> from server then a session is established...but i m unable to get the
> meaning of a dialog within a session...
> for a single call how many dialog can we have?? .
>
> do reply
>
>
> On Wed, Jun 25, 2014 at 11:41 AM, Kapoor, Geetika <geetika.kapoor at hp.com>
> wrote:
>
>>  Okay what I have is
>>
>>
>>
>> Sipp client        ||stateless proxy(designed by us)||        Sipp server
>>
>>
>>
>> My doubt if sipp can  work with stateless proxy. It works with statefull
>> proxy that I know.
>>
>>
>>
>> Thanks
>>
>> Geetika
>>
>>
>>
>> *From:* Shah Hussain Khattak [mailto:shahhusayn at msn.com]
>> *Sent:* Wednesday, June 25, 2014 11:31 AM
>> *To:* Kapoor, Geetika; discussion at sipforum.org
>> *Subject:* RE: [SIPForum-discussion] Sipp with stateless proxy
>>
>>
>>
>> Hi Sir,
>>
>>
>>
>> Can you please explain your question a little bit, what is the testing
>> scenario?
>>
>>
>>
>> Regards,
>>
>>
>> *An ounce of practice is worth more than tons of preaching. (Mohandas
>> Gandhi)*
>>
>> * "In three words I can sum up everything I've learned about life: it
>> goes on." (Robert Frost) **Shah Hussain*
>>  ------------------------------
>>
>> From: geetika.kapoor at hp.com
>> To: discussion at sipforum.org
>> Date: Mon, 23 Jun 2014 10:40:46 +0000
>> Subject: [SIPForum-discussion] Sipp with stateless proxy
>>
>> Can we use sip client with stateless proxy?
>>
>>
>> _______________________________________________ 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
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>>
>> _______________________________________________
>> 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
>>
>>
>
>
> --
> Aditya prakash(SDDE)
>
> _______________________________________________
> This is the SIP Forum discussion mailing list
> TO UNSUBSCRIBE, or edit your delivery options, please visit
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>
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