[OpenSIPS-Users] Issue with 'To' tag and t_reply

Thu Mar 26 14:56:21 EST 2020

Thanks a lot for your explanation, Ben! I thought that there can be an
issue with Server A not accepting  my new SIP response, it looks like
they're doing matching only based on SIP To tag and completely ignoring any
Call-ID or DID matching as well as From tag matching, in my case From tag
is always the same.

One more question, how do you think, can there be anything related with
topology hiding I'm using? I really doubt that but just in case...As far as
I understand my issue is not because I'm using topology hiding, but because
OpenSIPS first passes To tag from remote server and then generates one by
itself when I'm using a 't_reply' and Server A is just not accepting such
behaviour, trying to match any SIP responses to To tag passed in 183
Session Progress. I tried to change topology_hiding() to loose_route() and
nothing changed in my chain except for Server A now being able to see all
RRs and Vias inside my network.

Regards,
Yury.

пт, 27 мар. 2020 г. в 01:37, Yury Kirsanov <y.kirsanov at gmail.com>:

> But the question is still here - how can I send a different t_reply code
> from failure_route? And then stop processing any further SIP messages?
>
> пт, 27 мар. 2020 г. в 01:23, Yury Kirsanov <y.kirsanov at gmail.com>:
>
>> The problem is that I need to go through a list of SIP servers, analyze
>> response of each of them and if it's an error like 4XX, 5XX or 6XX I need
>> to send appropriate response to originating server. Let's say I'm not only
>> adding a Reason field but upon receipt of 404 Not Found I want to respond
>> with 480 Temporarily Unavailable with Reason: Q.850;Cause=41 for example?
>> But Server B first replied with 183 Session Progress playing back a message
>> 'Sorry, you need to top up your account' and then replied with SIP 402
>> Payment Required. I had to proxy 183 Session Progress back to Server A so
>> its SIP client could hear that message and then I'd like to signal 480
>> Temporarily Unavailable - but I can't as OpenSIPS is using completely
>> different To tag.
>>
>> I can't do this in onreply_route as I'm going through a list of SIP
>> servers (upstreams or downstreams), so it definitely needs to be done from
>> failure route, as far as I understand, and yes, I'm matching against 4XX,
>> 5XX and 6XX codes and I need to reply with 480 Temporarily Unavailable in
>> most cases so Server A would have a possibility to do failover to any other
>> server in that case. I don't want to just proxy 4XX, 5XX or 6XX response to
>> it.
>>
>> I've figured out why I have two 404 responses in my original call log - I
>> was using sl_send_reply instead of t_reply and it was using original To tag
>> but only on second attempt.
>>
>> Regards,
>> Yury.
>>
>> пт, 27 мар. 2020 г. в 01:02, Yury Kirsanov <y.kirsanov at gmail.com>:
>>
>>> Hi,
>>> As per my original email:
>>> 1. I was doing exactly as you suggested, in failure_route I'm using
>>> t_reply("404","Not Found") and it comes out with a wrong To: tag.
>>> 2. I don't need to proxy response from server B, I need to analyze its
>>> response and send a response to server A according to my needs.
>>>
>>> Currently it seems that t_reply is not using same To tag if 183 Session
>>> Progress has been proxied, which is strange as I have dialog running.
>>>
>>> Regards,
>>> Yury.
>>>
>>> чт, 26 мар. 2020 г. в 19:13, Yury Kirsanov <y.kirsanov at gmail.com>:
>>>
>>>> Hi,
>>>> I'm using an OpenSIPS as a proxy between two servers. First one is
>>>> sending SIP INVITE to OpenSIPS, then OpenSIPS forwards request to second
>>>> server. I'm creating a dialog on initial INVITE. Second server then replies
>>>> with SIP 183 Session Progress, plays back a message and then responds with
>>>> 4XX code, for example SIP 404 Not Found (indicating that number dialed is
>>>> disconnected). In OpenSIPS I'm receiving that reply and in failure_route
>>>> I'd like to change that code to a bit different SIP 404, so I'm using
>>>> following code:
>>>>
>>>> append_to_reply("Reason: Q.850;cause=1");
>>>> t_reply("404","Not Found");
>>>> exit;
>>>>
>>>> But in this case I can see that OpenSIPS generates additional branch
>>>> (??? not sure here) with different "To" tag and pushes it out and then
>>>> forwards original reply SIP packet even though I have an exit statement in
>>>> my failure_route. I tried to do sl_send_reply and behavior is pretty much
>>>> the same. Can someone let me know what I may be doing wrong? I need correct
>>>> "To" tag to be used (based on 183 Session Progress message from server B
>>>> and passed to server A previously) and second 404 shouldn't be forwarded
>>>> out.
>>>>
>>>> Here's an example of a call with my explanations
>>>>
>>>> Initial invite from server A, no 'to tag' as expected:
>>>>
>>>> INVITE sip:XXXXXXXXX at B.B.B.B SIP/2.0
>>>> Max-Forwards: 67
>>>> To: "XXXXXXXXX" <sip:XXXXXXXXX at B.B.B.B>
>>>> Call-ID: 469A5568-E092-4038-B1B8-13AC9B9571CA
>>>> Via: SIP/2.0/UDP A.A.A.A:5060;rport;branch=z9hG4bK773616538
>>>> From: "YYYYYYYYY" <sip:YYYYYYYYY at A.A.A.A>;tag=117583367
>>>> CSeq: 1741310 INVITE
>>>> User-Agent: User Agent
>>>> Contact: <sip:YYYYYYYYY at A.A.A.A:5060>
>>>> Allow: ACK, INVITE, BYE, CANCEL, REGISTER, REFER, OPTIONS, INFO,
>>>> SUBSCRIBE, NOTIFY
>>>> Date: Thu, 26 Mar 2020 07:54:55 GMT
>>>> Content-Type: application/sdp
>>>> Content-Length: 250
>>>>
>>>> v=0
>>>> o=dcom 1585209295 1585209295 IN IP4 A.A.A.A
>>>> s=SIP Call
>>>> c=IN IP4 A.A.A.A
>>>> t=0 0
>>>> m=audio 15340 RTP/AVP 8 0 18 101
>>>> a=rtpmap:8 PCMA/8000
>>>> a=rtpmap:0 PCMU/8000
>>>> a=rtpmap:18 G729/8000
>>>> a=fmtp:18 annexb=no
>>>> a=rtpmap:101 telephone-event/8000
>>>>
>>>> Response from OpenSIPS:
>>>>
>>>> SIP/2.0 100 Giving a try
>>>> To: "XXXXXXXXX" <sip:XXXXXXXXX at B.B.B.B>
>>>> Call-ID: 469A5568-E092-4038-B1B8-13AC9B9571CA
>>>> Via: SIP/2.0/UDP
>>>> A.A.A.A:5060;received=A.A.A.A;rport=5060;branch=z9hG4bK773616538
>>>> From: "YYYYYYYYY" <sip:YYYYYYYYY at A.A.A.A>;tag=117583367
>>>> CSeq: 1741310 INVITE
>>>> Server: Server Signature
>>>> Content-Length: 0
>>>>
>>>> OpenSIPS has forwarded packet to Server B and Server B responded with
>>>> 183 and assigned a 'To' tag:
>>>>
>>>> SIP/2.0 183 Session Progress
>>>> Via: SIP/2.0/UDP
>>>> A.A.A.A:5060;received=A.A.A.A;rport=5060;branch=z9hG4bK773616538
>>>> Call-ID: 469A5568-E092-4038-B1B8-13AC9B9571CA
>>>> From: "YYYYYYYYY" <sip:YYYYYYYYY at A.A.A.A>;tag=117583367
>>>> To: "XXXXXXXXX" <sip:XXXXXXXXX at B.B.B.B>;
>>>> *tag=0b49dc32-2c4b-413e-a349-c781a23d53b9*
>>>> CSeq: 1741310 INVITE
>>>> Server: PBX
>>>> Contact: <sip:B.B.B.B;did=d0a.99678f73>
>>>> Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE,
>>>> CANCEL, UPDATE, PRACK, REFER
>>>> Content-Type: application/sdp
>>>> Content-Length: 354
>>>>
>>>> v=0
>>>> o=- 1585209295 1585209297 IN IP4 B.B.B.B
>>>> s=Asterisk
>>>> c=IN IP4 B.B.B.B
>>>> t=0 0
>>>> a=rtpengine:673f999268ae
>>>> m=audio 32386 RTP/AVP 0 8 18 101
>>>> a=maxptime:150
>>>> a=rtpmap:0 PCMU/8000
>>>> a=rtpmap:8 PCMA/8000
>>>> a=rtpmap:18 G729/8000
>>>> a=rtpmap:101 telephone-event/8000
>>>> a=fmtp:18 annexb=no
>>>> a=fmtp:101 0-16
>>>> a=sendrecv
>>>> a=rtcp:32387
>>>> a=ptime:20
>>>>
>>>> Server B responds with SIP 404 after playing back message that number
>>>> is disconnected and I'm trying to reply to server A with custom Reason
>>>> message. To_tag is completely different from the To tag that has been
>>>> passed to server A after initial 183!!!
>>>>
>>>> SIP/2.0 404 Not Found
>>>> To: "XXXXXXXXX" <sip:XXXXXXXXX at B.B.B.B>;
>>>> *tag=a976.21514595b467be41a9b712a6b0b621d9*
>>>> Call-ID: 469A5568-E092-4038-B1B8-13AC9B9571CA
>>>> Via: SIP/2.0/UDP
>>>> A.A.A.A:5060;received=A.A.A.A;rport=5060;branch=z9hG4bK773616538
>>>> From: "YYYYYYYYY" <sip:YYYYYYYYY at A.A.A.A>;tag=117583367
>>>> CSeq: 1741310 INVITE
>>>> Reason: Q.850;cause=1;text="Number is disconnected"
>>>> Server: Server Signature
>>>> Content-Length: 0
>>>>
>>>> Of course, server A just ignores this message as it can't match 'To'
>>>> tag to its transaction. Now, for some reason, OpenSIPS forwards original
>>>> reply from Server B to Server A with the same 'To' tag as in 183 Session
>>>> Progress:
>>>>
>>>> SIP/2.0 404 Not Found
>>>> Via: SIP/2.0/UDP
>>>> A.A.A.A:5060;received=A.A.A.A;rport=5060;branch=z9hG4bK773616538
>>>> Call-ID: 469A5568-E092-4038-B1B8-13AC9B9571CA
>>>> From: "YYYYYYYYY" <sip:YYYYYYYYY at A.A.A.A>;tag=117583367
>>>> To: "XXXXXXXXX" <sip:XXXXXXXXX at B.B.B.B>;
>>>> *tag=0b49dc32-2c4b-413e-a349-c781a23d53b9*
>>>> CSeq: 1741310 INVITE
>>>> Server: PBX
>>>> Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE,
>>>> CANCEL, UPDATE, PRACK, REFER
>>>> Reason: Q.850;cause=1
>>>> Content-Length:  0
>>>>
>>>> And at this point Server A can match this reply and responds with an
>>>> ACK:
>>>>
>>>> ACK sip:XXXXXXXXX at B.B.B.B SIP/2.0
>>>> Via: SIP/2.0/UDP A.A.A.A:5060;rport;branch=z9hG4bK773616538
>>>> From: "YYYYYYYYY" <sip:YYYYYYYYY at A.A.A.A>;tag=117583367
>>>> To: "XXXXXXXXX" <sip:XXXXXXXXX at B.B.B.B>;
>>>> *tag=0b49dc32-2c4b-413e-a349-c781a23d53b9*
>>>> Call-ID: 469A5568-E092-4038-B1B8-13AC9B9571CA
>>>> CSeq: 1741310 ACK
>>>> Max-Forwards: 67
>>>> Contact: <sip:YYYYYYYYY at A.A.A.A:5060>
>>>> User-Agent: User Agent
>>>> Content-Length: 0
>>>>
>>>> I think that t_reply is creating a new transaction instead of using
>>>> existing one, but I'm not sure why and how to fix this?
>>>>
>>>> Thanks!
>>>>
>>>> Best regards,
>>>> Yury.
>>>>
>>>
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