[SIPForum-discussion] SIP Error 481

Fri Sep 28 01:41:37 UTC 2012

Team,

I'm currently studying the two attachments, requesting for your help guys if the solution was according to standards RFC 4028. I didn't see any Require:Timer.

>From header: 026571011;
To header: 026571011

Thank you very much.

Tristan Bulandus
Network Services Assurance
Customer Network, Network Engineer
0919-332-5174

PLDT Futsal Team Captain
[cid:image001.jpg at 01CD9D5D.093F5FC0]

From: Abhisek Acharya [mailto:abhisek.acharya at gmail.com]
Sent: Tuesday, September 25, 2012 4:21 PM
To: BULANDUS, Tristan L.
Cc: Abinash Sarangi; discussion at sipforum.org
Subject: Re: [SIPForum-discussion] SIP Error 481

yeah I guess
On Tue, Sep 25, 2012 at 12:44 PM, BULANDUS, Tristan L. <tlbulandus at pldt.com.ph<mailto:tlbulandus at pldt.com.ph>> wrote:
Is that the same thing as the Require:timer as suggested before?

From: Abhisek Acharya [mailto:abhisek.acharya at gmail.com<mailto:abhisek.acharya at gmail.com>]
Sent: Tuesday, September 25, 2012 3:12 PM

To: BULANDUS, Tristan L.
Cc: Abinash Sarangi; discussion at sipforum.org<mailto:discussion at sipforum.org>
Subject: Re: [SIPForum-discussion] SIP Error 481

here we go...I got it now...SessionExpiry mechanism was the culprit

Regards
Abhisek Acharya
GS_LAB
PUNE

On Tue, Sep 25, 2012 at 11:50 AM, BULANDUS, Tristan L. <tlbulandus at pldt.com.ph<mailto:tlbulandus at pldt.com.ph>> wrote:
Hi Abhisek,

IPPBX guy says they use version 10 which is the highest. They said that they support RFC 4028.
They disabled the session-timers and they reported that call does not get cut after 15 minutes.

I'm asking for the new pcap so that I can validate if our analysis is confirmed.

Thank you so much for your help.

From: Abhisek Acharya [mailto:abhisek.acharya at gmail.com<mailto:abhisek.acharya at gmail.com>]
Sent: Tuesday, September 25, 2012 2:16 PM

To: BULANDUS, Tristan L.
Cc: Abinash Sarangi; discussion at sipforum.org<mailto:discussion at sipforum.org>
Subject: Re: [SIPForum-discussion] SIP Error 481

Hey Bulandus,

Now I am really eager to see what the IPPBX guy says.As you are in a good position to describe about the problem and product behavior.Correct me if I am wrong.

Regards
Abhisek Acharya
On Tue, Sep 25, 2012 at 1:52 AM, BULANDUS, Tristan L. <tlbulandus at pldt.com.ph<mailto:tlbulandus at pldt.com.ph>> wrote:
Thanks Abinash,

For option 1 - I don't think this will work since its stated that this is the default seting, assuming that my client isn't that techy at all he wouldn't have changed the sipconf settings which means the default setting doesn't work.

For option 2 - It only says about asterisk sending re-invites periodically to remote endpoints, however it doesn't say anything about Require:Timer

For option 3 - I don't think I need a 420 Bad extension.

I read on one website though that some Asterisk IPPBXs support RFC 4028 while others do, particularly only Asterisk version 1.6 does, from what I read.
Thank you very much.

Tristan Bulandus
Network Products & Services Design
Cadet Network Engineer
Batch RING
________________________________________
From: Abinash Sarangi [s_abinash at hotmail.com<mailto:s_abinash at hotmail.com>]
Sent: Monday, September 24, 2012 8:48 PM
To: Abhisek Acharya
Cc: BULANDUS, Tristan L.; discussion at sipforum.org<mailto:discussion at sipforum.org>
Subject: RE: [SIPForum-discussion] SIP Error 481
Hi Bulandus,
 some information regarding asterisk session timer support

The Session-Timers can be configured on a system-wide, per-user, or per-peer basis. The peruser/ per-peer settings override the global settings. The following new parameters have been added to the sip.conf file. session-timers=["accept", "originate", "refuse"] session-expires=[integer] session-minse=[integer] session-refresher=["uas", "uac"]
The session-timers parameter in sip.conf defines the mode of operation of SIP session-timers feature in Asterisk. The Asterisk can be configured in one of the following three modes:
1. Accept :: In the "accept" mode, the Asterisk server honors session-timers requests<http://www.asterisk.org/doxygen/trunk/structrequests.html> made by remote end-points. A remote end-point can request Asterisk to engage session-timers by either sending it an INVITE request with a "Supported: timer" header in it or by responding to Asterisk's INVITE with a 200 OK that contains Session-Expires: header in it. In this mode, the Asterisk server does not request session-timers from remote end-points. This is the default mode. 2. Originate :: In the "originate" mode, the Asterisk server requests<http://www.asterisk.org/doxygen/trunk/structrequests.html> the remote end-points to activate session-timers in addition to honoring such requests<http://www.asterisk.org/doxygen/trunk/structrequests.html> made by the remote end-pints. In order to get as much protection as possible against hanging SIP channels due to network or end-point failures, Asterisk resends periodic re-INVITEs even if a remote end-point does not support the session-timers feature. 3. Refuse :: In the "refuse" mode, Asterisk acts as if it does not support session- timers for inbound or outbound requests<http://www.asterisk.org/doxygen/trunk/structrequests.html>. If a remote end-point requests<http://www.asterisk.org/doxygen/trunk/structrequests.html> session-timers in a dialog, then Asterisk ignores that request unless it's noted as a requirement (Require: header), in which case the INVITE is rejected with a 420 Bad Extension response.

Thanks
-Abinash

________________________________
Date: Mon, 24 Sep 2012 16:43:45 +0530
Subject: Re: [SIPForum-discussion] SIP Error 481
From: abhisek.acharya at gmail.com<mailto:abhisek.acharya at gmail.com>
To: s_abinash at hotmail.com<mailto:s_abinash at hotmail.com>
CC: tlbulandus at pldt.com.ph<mailto:tlbulandus at pldt.com.ph>; discussion at sipforum.org<mailto:discussion at sipforum.org>

Hardfought issue fellas.
On Mon, Sep 24, 2012 at 4:41 PM, Abinash Sarangi <s_abinash at hotmail.com<mailto:s_abinash at hotmail.com><mailto:s_abinash at hotmail.com<mailto:s_abinash at hotmail.com>>> wrote:
Hi Bulandus,

Have one question. In case of alcatel IP PBX call, whose is this IP (10.249.211.5)..is it behind 10.249.211.20 - Acme SBC facing Alcatel
IPPBX

I agree with abhisek on session timers part.. As I told you earlier, make some changes in sip.conf and see how the active calls are getting affected.

Thanks
-Abinash

________________________________
Date: Mon, 24 Sep 2012 14:22:45 +0530

Subject: Re: [SIPForum-discussion] SIP Error 481
From: abhisek.acharya at gmail.com<mailto:abhisek.acharya at gmail.com><mailto:abhisek.acharya at gmail.com<mailto:abhisek.acharya at gmail.com>>
To: tlbulandus at pldt.com.ph<mailto:tlbulandus at pldt.com.ph><mailto:tlbulandus at pldt.com.ph<mailto:tlbulandus at pldt.com.ph>>
CC: s_abinash at hotmail.com<mailto:s_abinash at hotmail.com><mailto:s_abinash at hotmail.com<mailto:s_abinash at hotmail.com>>; discussion at sipforum.org<mailto:discussion at sipforum.org><mailto:discussion at sipforum.org<mailto:discussion at sipforum.org>>

Hey Bulandus,

If any IPPBX says it supports RFC4028 and session-refresher mechanism then it MUST send Reuire:timer to kick in the Session-Expiry process.Also please check in the IPPBx if they have any relevant settings and all.

Regards
Abhisek Acharya
GS-LAB,PUNE
INDIA
On Mon, Sep 24, 2012 at 2:10 PM, BULANDUS, Tristan L. <tlbulandus at pldt.com.ph<mailto:tlbulandus at pldt.com.ph><mailto:tlbulandus at pldt.com.ph<mailto:tlbulandus at pldt.com.ph>>> wrote:
Thanks Buddy,
I found what you're talking about on Require:timer,

Before I get back to my client using Asterisk IPPBX, may I ask what possible equivalent configurations can they do on their side to put this - "Require:timer" ?

Thank you very much.

Tristan Bulandus
Network Services Assurance
Customer Network, Network Engineer
0919-332-5174

PLDT Futsal Team Captain
[cid:image001.jpg at 01CD9A73.600B1CF0]<mailto:[cid:image001.jpg at 01CD9A73.600B1CF0]>

From: Abhisek Acharya [mailto:abhisek.acharya at gmail.com<mailto:abhisek.acharya at gmail.com><mailto:abhisek.acharya at gmail.com<mailto:abhisek.acharya at gmail.com>>]
Sent: Monday, September 24, 2012 4:33 PM
To: BULANDUS, Tristan L.
Cc: Abinash Sarangi; discussion at sipforum.org<mailto:discussion at sipforum.org><mailto:discussion at sipforum.org<mailto:discussion at sipforum.org>>

Subject: Re: [SIPForum-discussion] SIP Error 481
Hey buddy,

dont go for 3262 as it is for PRACK.Go for RFC4028.

If you check the succesfull call and filter the call with call-id (sip.Call-ID == "gfcfgejoidomflpfslsscslled3eidc2 at SoftX3000")then you can find the succesfull call between the Alcatel PBX 10.10.40.5 and 10.249.211.5 where you can see
although first 200 OK from alcatel PBX does not contain Require:timer but subsequent refrehser requests 200 OK response contains the Require:timer value and call went on
succesfully.You can see RE-INVITES going from 10.10.40.5 IP and getting 200 OK response with Require:timer value.Thsi is as per standard and call is okie.
Even if there are few strnage things in sucesfull call still then I will give thumbs up to the call.

UNSUCCESFULL CALL

If you check the unsuccesfull call (sip.Call-ID == "mnpedempgcnomejfosjoekm2sfnnmdfo at SoftX3000")then you can find lot of session refreshers are going from 10.251.3.153 and also lot of session refreshers are coming to 10.251.3.153 and all these are
happening without being acknolwedged.Thsi is lot of ambigousness in the system and finally call drops.

Can you please by any chance completely disable Session-Refresher mechanism for some time and test.
On Mon, Sep 24, 2012 at 1:55 PM, BULANDUS, Tristan L. <tlbulandus at pldt.com.ph<mailto:tlbulandus at pldt.com.ph><mailto:tlbulandus at pldt.com.ph<mailto:tlbulandus at pldt.com.ph>>> wrote:
My observation is that Alcatel IPPBX contains 100rel in its 200 OK message whereas the Asterisk does not contain this. Could this be the reason.

I read RFC 3262 although will need some help in understanding it.

Thank you very much.

Tristan Bulandus
Network Services Assurance
Customer Network, Network Engineer
0919-332-5174

PLDT Futsal Team Captain
[cid:image001.jpg at 01CD9A73.600B1CF0]<mailto:[cid:image001.jpg at 01CD9A73.600B1CF0]>

From: Abhisek Acharya [mailto:abhisek.acharya at gmail.com<mailto:abhisek.acharya at gmail.com><mailto:abhisek.acharya at gmail.com<mailto:abhisek.acharya at gmail.com>>]
Sent: Monday, September 24, 2012 4:08 PM
To: BULANDUS, Tristan L.
Cc: Abinash Sarangi

Subject: Re: [SIPForum-discussion] SIP Error 481

I checked both the traces and I will get back to you.My initial reaction is its a Session-Refresher problem .I will send you my analysis
On Mon, Sep 24, 2012 at 12:45 PM, BULANDUS, Tristan L. <tlbulandus at pldt.com.ph<mailto:tlbulandus at pldt.com.ph><mailto:tlbulandus at pldt.com.ph<mailto:tlbulandus at pldt.com.ph>>> wrote:
Resending, while still awaiting approval for attachment from moderator.

Thank you very much.

Tristan Bulandus
Network Services Assurance
Customer Network, Network Engineer
0919-332-5174

PLDT Futsal Team Captain
[cid:image001.jpg at 01CD9A73.600B1CF0]<mailto:[cid:image001.jpg at 01CD9A73.600B1CF0]>

From: BULANDUS, Tristan L.
Sent: Monday, September 24, 2012 3:12 PM
To: 'Abhisek Acharya'; Abinash Sarangi
Cc: discussion at sipforum.org<mailto:discussion at sipforum.org><mailto:discussion at sipforum.org<mailto:discussion at sipforum.org>>

Subject: RE: [SIPForum-discussion] SIP Error 481
Team,

Please see results:

A. Unsuccessful - gets cut after 5-10 minutes
10.251.4.153 - Asterisk IPPBX
10.249.211.20 - Acme SBC facing Asterisk IPPBX
10.249.212.32 - Acme SBC facing Huawei Softswitch
10.249.200.1 - Huawei Softswitch facing PSTN

B. Successful -  did not get cut after 17 minutes
10.10.40.5 - Alcatel IPPBX
10.249.211.20 - Acme SBC facing Alcatel IPPBX

Thank you very much.

Tristan Bulandus
Network Services Assurance
Customer Network, Network Engineer
0919-332-5174

PLDT Futsal Team Captain
[cid:image001.jpg at 01CD9A73.600B1CF0]<mailto:[cid:image001.jpg at 01CD9A73.600B1CF0]>

From: Abhisek Acharya [mailto:abhisek.acharya at gmail.com<mailto:abhisek.acharya at gmail.com><mailto:abhisek.acharya at gmail.com<mailto:abhisek.acharya at gmail.com>>]
Sent: Monday, September 24, 2012 2:59 PM
To: Abinash Sarangi
Cc: BULANDUS, Tristan L.; discussion at sipforum.org<mailto:discussion at sipforum.org><mailto:discussion at sipforum.org<mailto:discussion at sipforum.org>>

Subject: Re: [SIPForum-discussion] SIP Error 481
Hey Abinash,

Just one quick question before we check the conf file.I check in the trace the UAS replies with Supported:timer where as UAC sends supported:timer to show the support for the extension.Dont you think the IPPBX is violating the standard?

Regards
Abhisek Acharya
On Mon, Sep 24, 2012 at 11:35 AM, Abinash Sarangi <s_abinash at hotmail.com<mailto:s_abinash at hotmail.com><mailto:s_abinash at hotmail.com<mailto:s_abinash at hotmail.com>>> wrote:
Hi Bulandus,

can you check configuring  session-timers=accept in sip.conf of your asterisk server and in case there is any other value to set a higher session-expires value..plz share the traces with asterisk and alcatel IP PBX

Thanks
-Abinash
> From: tlbulandus at pldt.com.ph<mailto:tlbulandus at pldt.com.ph><mailto:tlbulandus at pldt.com.ph<mailto:tlbulandus at pldt.com.ph>>
> To: abhisek.acharya at gmail.com<mailto:abhisek.acharya at gmail.com><mailto:abhisek.acharya at gmail.com<mailto:abhisek.acharya at gmail.com>>
> Date: Sat, 22 Sep 2012 10:28:24 +0800
> CC: discussion at sipforum.org<mailto:discussion at sipforum.org><mailto:discussion at sipforum.org<mailto:discussion at sipforum.org>>

> Subject: Re: [SIPForum-discussion] SIP Error 481
>
> Hi Abhisek,
>
> Thank you very much for the expertise that you are unconditionally sharing. However, I have asked some of my guys to test long duration calls using the Alcatel IPPBX, specifically Omni-PCX and the report I got from them is that calls are not cut after 5-10 minutes.
>
> My only problem is that when IPPBX used to connect to our network is the Asterisk, it gets cut after approximately 5 minutes. The trace I sent previously wherein I used an Alcatel IPPBX was actually intentionally cancelled during the time of test just to show that ringback tone would be successful. I can get an equivalent trace for a successful call that does not get cut off next week, and I will quickly post next week.
>
> Now this is the question that keeps on boggling my head, why doesn't it get cut using an Alcatel IPPBX, I've searched the pcap trace and I can't find any "Required:timer"
>
> For your comments please.
>
> Thank you very much.
>
> Tristan Bulandus
> Network Products & Services Design
> Cadet Network Engineer
> Batch RING
> ________________________________________
> From: Abhisek Acharya [abhisek.acharya at gmail.com<mailto:abhisek.acharya at gmail.com><mailto:abhisek.acharya at gmail.com<mailto:abhisek.acharya at gmail.com>>]
> Sent: Friday, September 21, 2012 7:10 PM
> To: BULANDUS, Tristan L.
> Cc: discussion at sipforum.org<mailto:discussion at sipforum.org><mailto:discussion at sipforum.org<mailto:discussion at sipforum.org>>
> Subject: Re: [SIPForum-discussion] SIP Error 481
>
> Hey Bulandus,
>
> In the mean while can you ask the IP-PBX guys and let them know that IP-PBX is not sending Require:timer even if it supports Session-Expiry mechanism.I am sure they will be happy to know that you raised this issue.Awaiting for your reply over the same.
>
>
> Regards
> GS-LAB-PVTLTD,PUNE
> Abhisek Acharya
>
> On Fri, Sep 21, 2012 at 4:37 PM, Abhisek Acharya <abhisek.acharya at gmail.com<mailto:abhisek.acharya at gmail.com><mailto:abhisek.acharya at gmail.com<mailto:abhisek.acharya at gmail.com>><mailto:abhisek.acharya at gmail.com<mailto:abhisek.acharya at gmail.com><mailto:abhisek.acharya at gmail.com<mailto:abhisek.acharya at gmail.com>>>> wrote:
> Hey
>
> I checkd the trace that you sent and this is a failed call.Call is being CANCELLED by the calling party here.
>
> Regards
> Abhisek Acharya
>
>
> On Fri, Sep 21, 2012 at 4:35 PM, Abhisek Acharya <abhisek.acharya at gmail.com<mailto:abhisek.acharya at gmail.com><mailto:abhisek.acharya at gmail.com<mailto:abhisek.acharya at gmail.com>><mailto:abhisek.acharya at gmail.com<mailto:abhisek.acharya at gmail.com><mailto:abhisek.acharya at gmail.com<mailto:abhisek.acharya at gmail.com>>>> wrote:
> Hi Dear Bulandus,
>
> Low session-Expiry values are not dangerous but this can create a lot of mid-dialog request and responses which will unnecessarily over load the network.
> Session-Expiry values should be greater than 90 seconds.It should not be less than this.
>
> Significance of Require:timer
>
> 1.Let us assume UAC sends supported:timer that means it supports session-expiry mechanism.If the UAS supports Session-Refresher mechanism then it MUST send Require:timer to kick in the Session-Refresher mechanism.It does not matter who is the refresher the UAS MUST send Require:timer to kick in the Session-Expiry method.
> I am yet to check the RFC.Let me check RFC 4028 and will get back to you.Also I will check your new trace and will get back to you as well.
>
> Regards
> Abhisek Acharya
> GS-LAB,PVT LTD,
> PUNE
>
>
>
>
> On Fri, Sep 21, 2012 at 1:32 PM, BULANDUS, Tristan L. <tlbulandus at pldt.com.ph<mailto:tlbulandus at pldt.com.ph><mailto:tlbulandus at pldt.com.ph<mailto:tlbulandus at pldt.com.ph>><mailto:tlbulandus at pldt.com.ph<mailto:tlbulandus at pldt.com.ph><mailto:tlbulandus at pldt.com.ph<mailto:tlbulandus at pldt.com.ph>>>> wrote:
> Attachment I have here represents an Alcatel IPPBX with almost the same scenario but call does not get cut.
>
> For your comments please.
>
> Thank you very much.
>
> Tristan Bulandus
> Network Services Assurance
> Customer Network, Network Engineer
> 0919-332-5174
>
> PLDT Futsal Team Captain
> [cid:image001.jpg at 01CD9812.7D48AAC0]<mailto:[cid:image001.jpg at 01CD9812.7D48AAC0]><mailto:[cid:image001.jpg at 01CD9812.7D48AAC0]>
>
> From: BULANDUS, Tristan L.
> Sent: Friday, September 21, 2012 3:48 PM
> To: 'Abhisek Acharya'
> Cc: discussion at sipforum.org<mailto:discussion at sipforum.org><mailto:discussion at sipforum.org<mailto:discussion at sipforum.org>><mailto:discussion at sipforum.org<mailto:discussion at sipforum.org><mailto:discussion at sipforum.org<mailto:discussion at sipforum.org>>>
> Subject: RE: [SIPForum-discussion] SIP Error 481
>
> What can "Require:timer" do?
>
> Is it really dangerous to have a low session-expiration timer?
>
> Thank you very much.
>
> Tristan Bulandus
> Network Services Assurance
> Customer Network, Network Engineer
> 0919-332-5174
>
> PLDT Futsal Team Captain
> [cid:image001.jpg at 01CD9812.7D48AAC0]<mailto:[cid:image001.jpg at 01CD9812.7D48AAC0]><mailto:[cid:image001.jpg at 01CD9812.7D48AAC0]>
>
> From: Abhisek Acharya [mailto:abhisek.acharya at gmail.com<mailto:abhisek.acharya at gmail.com><mailto:abhisek.acharya at gmail.com<mailto:abhisek.acharya at gmail.com>><mailto:abhisek.acharya at gmail.com<mailto:abhisek.acharya at gmail.com><mailto:abhisek.acharya at gmail.com<mailto:abhisek.acharya at gmail.com>>>]
> Sent: Friday, September 21, 2012 12:44 PM
>
> To: BULANDUS, Tristan L.
> Cc: discussion at sipforum.org<mailto:discussion at sipforum.org><mailto:discussion at sipforum.org<mailto:discussion at sipforum.org>><mailto:discussion at sipforum.org<mailto:discussion at sipforum.org><mailto:discussion at sipforum.org<mailto:discussion at sipforum.org>>>
> Subject: Re: [SIPForum-discussion] SIP Error 481
>
> Hey BULANDUS,
>
> If I am not wrong Your call gets disconnected in every five minutes.If that is true then in your system session refresher mechanism is not working as per the standard.
>
> I saw a INVITE is coming from 10.249.211.20 with Supported:timer value and Session-Expiry time with 300 seconds.And in 200 OK response the 10.251.3.153 MUST send Require:timer but rather it is sending Supported timer which is not correct.First thing what you need to check with the AsteriskIP-PBX is why it is not sending Require:timer .May be you can raise a BUG or something.
>
> Post that I am seeing a lot of Session-Refresher requests from 10.251.3.153 IP travelling till the end.And at times it is not getting the proper response and and request is pending on the other side.Because the INVITE is not processed and yet another comes.
>
> And Also 10.251.3.153 sending so many INVITES with chaning the sdp body without getting the response is also a problem.
> And due to some reason 10.249.211.20 removes the state of the dialog which it should not do.
>
> FEW SUGGESTIONS
>
> 1.Check Why 10.251.3.153 not replying with Reuire:timer even if it supports Session-refresher mechanism.It should not send Supported:timer rather it should send Require:timer value.
>
> 2.Change the SessionExpiry value to a big number such as 1800 seconds or something becuase small session-expiry values are dangerous to network as it will create a lot of congestion with huge number of message flow.
>
> 3.Calls getting disconnected excatly in 5 minutes because Session-Refresher value is 300 seconds.481 response has nothing to do with the call disconnection as this happens way after the call set up.
>
> Awaiting for your response over the same.
>
> Regards
> Abhisek Acharya
> GS-LAB
> PUNE
>
>
>
>
>
> Please feel free to ask queries if any or if any something is not clear on this.
>
>
>
>
>
>
>
>
> On Fri, Sep 21, 2012 at 6:36 AM, BULANDUS, Tristan L. <tlbulandus at pldt.com.ph<mailto:tlbulandus at pldt.com.ph><mailto:tlbulandus at pldt.com.ph<mailto:tlbulandus at pldt.com.ph>><mailto:tlbulandus at pldt.com.ph<mailto:tlbulandus at pldt.com.ph><mailto:tlbulandus at pldt.com.ph<mailto:tlbulandus at pldt.com.ph>>>> wrote:
> Team,
> Appreciate help on below.
>
> 10.251.3.153 - Asterisk IPPBX
> 10.249.211.20 - Acme SBC facing Asterisk
> 10.249.212.32 - Acme SBC facing Huawei Softswitch
> 10.249.200.1 - Huawei Softswitch facing PSTN
>
> Thank you very much.
>
> Tristan Bulandus
> Network Services Assurance
> Customer Network, Network Engineer
> 0919-332-5174
>
> PLDT Futsal Team Captain
> [cid:image001.jpg at 01CD9812.7D48AAC0]<mailto:[cid:image001.jpg at 01CD9812.7D48AAC0]><mailto:[cid:image001.jpg at 01CD9812.7D48AAC0]>
>
> From: Abhisek Acharya [mailto:abhisek.acharya at gmail.com<mailto:abhisek.acharya at gmail.com><mailto:abhisek.acharya at gmail.com<mailto:abhisek.acharya at gmail.com>><mailto:abhisek.acharya at gmail.com<mailto:abhisek.acharya at gmail.com><mailto:abhisek.acharya at gmail.com<mailto:abhisek.acharya at gmail.com>>>]
> Sent: Thursday, September 20, 2012 12:37 PM
> To: BULANDUS, Tristan L.
> Cc: discussion at sipforum.org<mailto:discussion at sipforum.org><mailto:discussion at sipforum.org<mailto:discussion at sipforum.org>><mailto:discussion at sipforum.org<mailto:discussion at sipforum.org><mailto:discussion at sipforum.org<mailto:discussion at sipforum.org>>>
> Subject: Re: [SIPForum-discussion] SIP Error 481
>
> Hey Buddy,
>
>
> My initial question to you is open the INVITE message and check the FROM and TO header.Also try to check the TO header in specific.There should be no TO-TAG present in the TO header.As the Initial INVITE never contains a TO-TAG.
>
>
> 481 means call-leg does not exist.You are saying call gets disconnected after 5 minutes.Which means the call gets disconnected after the session is established.Can you please check if the RE-INVITE contains the same FROM and TO TAG or not.
>
> Also Please send me the full wireshark trace and IP addresses.So that I can quickly check and get back to you.
>
> Regards
> Abhisek
> On Wed, Sep 19, 2012 at 2:03 PM, BULANDUS, Tristan L. <tlbulandus at pldt.com.ph<mailto:tlbulandus at pldt.com.ph><mailto:tlbulandus at pldt.com.ph<mailto:tlbulandus at pldt.com.ph>><mailto:tlbulandus at pldt.com.ph<mailto:tlbulandus at pldt.com.ph><mailto:tlbulandus at pldt.com.ph<mailto:tlbulandus at pldt.com.ph>>>> wrote:
> Hi Team,
>
> We have a network set-up of SIP Peering services, with details below:
>
> ASTERISK PBX - SESSION BORDER CONTROLLER (PROXY) - HUAWEI SOFTSWITCH (PSTN GATEWAY)
> [cid:image002.jpg at 01CD9812.7D48AAC0]<mailto:[cid:image002.jpg at 01CD9812.7D48AAC0]><mailto:[cid:image002.jpg at 01CD9812.7D48AAC0]>
>
> Calls originating from the PSTN Gateway is able to establish connection to Asterisk PBX but the call gets cut after 5 minutes. We experience Error 481 from (SBC - 10.249.211.20) to (Asterisk - 10.251.3.153).
>
> Expert advice please on how to resolve this.
>
> Thank you very much.
>
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>
>
>
>
>
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