Server is a Opensips server 1.10.0-tls (linux). It can handle conversations to/from local stations and recently it has been updated to allow stations from external systems. It does this by changing the username, ip and port in the $ru if the station doesn't exist locally ($tu is untouched). This works fine for invites, calls and all similar messages.
The problem I'm having is that byes coming from the external server, that are being passed on to the local client station, are being rejected by a 481 (call leg/transaction does not exist) that i can confirm is coming from the client software, yet it accepts byes from local stations on the same server without any bother. Calls from local to local end ok, calls from local to external all shutdown ok, it's just calls from external caller to local callee that won't close (it is the callee that is saying 481).
I understand this is caused by transaction matching not occurring due to something different in the tags in the to/from and the call-id; i understand that something (like my $ru script) that changes parts of the $ru might have an effect on the hash to determine the transaction, but i don't change the tags or the callid, just the $ru name to make it go to the right IP and station name.
My question is how do i go about solving this on the server without changing the client applications? I've included some examples of the messages being sent below taken from a wireshark capture on the client workstation, so i'm unsure what i'm doing wrong..I've tried different things on the server with no luck. Is there a way to mark or tell the client via sip messages to shutdown the conversation regardless of transaction matching?
I would greatly appreciate any aid in this as I've been pulling my hair out about this for some time.
Message examples for External caller (103 on server 5.44) to Local Callee (local name is wks2, external ref name is 155, on server 3.3, client is 3.0). First bye is the problem, second bye is me closing the hanging connection on the client.
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INVITE sip:[email protected]:5050 SIP/2.0
Record-Route: <sip:172.16.3.3;lr;ftag=as678f227c>
Via: SIP/2.0/UDP 172.16.3.3:5060;branch=z9hG4bKd066.93df6e05.0
Via: SIP/2.0/UDP 172.16.5.44:5060;received=172.16.5.44;branch=z9hG4bK50029c58;rport=5060
Max-Forwards: 69
From: "Station 103 " <sip:[email protected]>;tag=as678f227c
To: <sip:[email protected]:5060>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.13.1~dfsg1-3+deb7u3
Date: Thu, 25 Jun 2015 14:27:31 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 296
Redirect-to: sip:[email protected]:5060
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SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 172.16.3.3:5060;branch=z9hG4bKd066.93df6e05.0;received=172.16.3.3;rport=5060
Via: SIP/2.0/UDP 172.16.5.44:5060;received=172.16.5.44;branch=z9hG4bK50029c58;rport=5060
To: <sip:[email protected]:5060>;tag=1262186908
From: "Station 103 " <sip:[email protected]>;tag=as678f227c
Call-ID: [email protected]:5060
CSeq: 102 INVITE
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, SUBSCRIBE
Content-Length: 0
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SIP/2.0 200 Ok
Via: SIP/2.0/UDP 172.16.3.3:5060;branch=z9hG4bKd066.93df6e05.0;received=172.16.3.3;rport=5060
Via: SIP/2.0/UDP 172.16.5.44:5060;received=172.16.5.44;branch=z9hG4bK50029c58;rport=5060
To: <sip:[email protected]:5060>;tag=1262186908
From: "Station 103 " <sip:[email protected]>;tag=as678f227c
Call-ID: [email protected]:5060
CSeq: 102 INVITE
Contact: <sip:172.16.3.0:5050>
Record-Route: <sip:172.16.3.3;lr;ftag=as678f227c>
Server: www.sipsorcery.com
Content-Length: 161
Content-Type: application/sdp
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ACK sip:172.16.3.0:5050 SIP/2.0
Via: SIP/2.0/UDP 172.16.3.3:5060;branch=z9hG4bKd066.93df6e05.2
Via: SIP/2.0/UDP 172.16.5.44:5060;received=172.16.5.44;branch=z9hG4bK49d179f0;rport=5060
Max-Forwards: 69
From: "Station 103 " <sip:[email protected]>;tag=as678f227c
To: <sip:[email protected]:5060>;tag=1262186908
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.13.1~dfsg1-3+deb7u3
Content-Length: 0
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BYE sip:172.16.3.0:5050 SIP/2.0
Via: SIP/2.0/UDP 172.16.3.3:5060;branch=z9hG4bKe066.64948323.0
Via: SIP/2.0/UDP 172.16.5.44:5060;received=172.16.5.44;branch=z9hG4bK1e381a63;rport=5060
Max-Forwards: 69
From: "Station 103 " <sip:[email protected]>;tag=as678f227c
To: <sip:[email protected]:5060>;tag=1262186908
Call-ID: [email protected]:5060
CSeq: 103 BYE
User-Agent: Asterisk PBX 1.8.13.1~dfsg1-3+deb7u3
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
----------
SIP/2.0 481 CallLegTransactionDoesNotExist
Via: SIP/2.0/UDP 172.16.3.3:5060;branch=z9hG4bKe066.64948323.0;received=172.16.3.3;rport=5060
Via: SIP/2.0/UDP 172.16.5.44:5060;received=172.16.5.44;branch=z9hG4bK1e381a63;rport=5060
To: <sip:[email protected]:5060>;tag=1262186908
From: "Station 103 " <sip:[email protected]>;tag=as678f227c
Call-ID: [email protected]:5060
CSeq: 103 BYE
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, SUBSCRIBE
Content-Length: 0
----------
BYE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.3.0:5050;branch=z9hG4bKfe13e63d99524b06846bde0fedbd8a69;rport
To: "Station 103 " <sip:[email protected]>;tag=as678f227c
From: <sip:[email protected]:5060>;tag=1262186908
Call-ID: [email protected]:5060
CSeq: 103 BYE
Max-Forwards: 70
Route: <sip:172.16.3.3;lr;ftag=as678f227c>
Content-Length: 0
----------
SIP/2.0 481 Call leg/transaction does not exist
Via: SIP/2.0/UDP 172.16.3.0:5050;received=172.16.3.0;branch=z9hG4bKfe13e63d99524b06846bde0fedbd8a69;rport=5050
From: <sip:[email protected]:5060>;tag=1262186908
To: "Station 103 " <sip:[email protected]>;tag=as678f227c
Call-ID: [email protected]:5060
CSeq: 103 BYE
Server: Asterisk PBX 1.8.13.1~dfsg1-3+deb7u3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
----------
Thank you :-)
