Hi Nick,
The BYE is not properly formed and rejected by script - in the 200 OK of
the INVITE, you can see that your opensips is doing Record-Routing, but
the BYE does not contain the corresponding Route hdr, so SIP routing is
impossible.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 04/09/2013 08:05 PM, Nick Khamis wrote:
Hello Everyone,
I saw an earlier post about this issue:
http://www.mail-archive.com/[email protected]/msg23052.html
And was wondering if there was anything we can do on our end to fix
this problem? It seems that providers are not obligated to maintain
RR? When the caller (internal) initiates the BYE everything is ok, but
not the case when the callee (external) initiates the BYE.
192.168.2.5 <http://192.168.2.5>: OpenSIPS
192.168.2.10 <http://192.168.2.10>: Asterisk
70.10.163.44 <http://70.10.163.44>: Public IP
108.59.2.133 <http://108.59.2.133>: Service Provider
U 2013/04/09 12:17:02.920454 192.168.2.10:5060
<http://192.168.2.10:5060> -> 192.168.2.5:5060 <http://192.168.2.5:5060>
SIP/2.0 200 OK.
Via: SIP/2.0/UDP
192.168.2.5;branch=z9hG4bKac2e.554c6e93.0;received=192.168.2.5;rport=5060.
Via: SIP/2.0/UDP
192.168.2.11:5060;rport=5060;received=192.168.2.11;branch=z9hG4bK42f3f16e7BC15DF1.
Record-Route: <sip:192.168.2.5;lr;did=392.62562fb2>.
From: "1001" <sip:[email protected]
<mailto:sip%[email protected]>>;tag=FCA0BFC0-B585477D.
To: <sip:[email protected]
<mailto:sip%[email protected]>;user=phone>;tag=as0a76fcde.
Call-ID: [email protected]
<mailto:[email protected]>.
CSeq: 1 INVITE.
Server: Asterisk PBX UNKNOWN__and_probably_unsupported.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH.
Supported: replaces, timer.
Contact: <sip:[email protected]:5060
<http://sip:[email protected]:5060>>.
Content-Type: application/sdp.
Content-Length: 312.
.
v=0.
o=root 1860889533 1860889534 IN IP4 192.168.2.10.
s=Asterisk PBX UNKNOWN__and_probably_unsupported.
c=IN IP4 192.168.2.10.
t=0 0.
m=audio 60646 RTP/AVP 18 101.
a=rtpmap:18 G729/8000.
a=fmtp:18 annexb=no.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=silenceSupp:off - - - -.
a=ptime:20.
a=sendrecv.
ACC: transaction answered:
timestamp=1365524222;method=INVITE;from_tag=FCA0BFC0-B585477D;to_tag=as0a76fcde;[email protected]
<mailto:[email protected]>;code=200;reason=OK
U 2013/04/09 12:17:02.939608 192.168.2.5:5060
<http://192.168.2.5:5060> -> 192.168.2.11:5060 <http://192.168.2.11:5060>
SIP/2.0 200 OK.
Via: SIP/2.0/UDP
192.168.2.11:5060;rport=5060;received=192.168.2.11;branch=z9hG4bK42f3f16e7BC15DF1.
Record-Route: <sip:192.168.2.5;lr;did=392.62562fb2>.
From: "1001" <sip:[email protected]
<mailto:sip%[email protected]>>;tag=FCA0BFC0-B585477D.
To: <sip:[email protected]
<mailto:sip%[email protected]>;user=phone>;tag=as0a76fcde.
Call-ID: [email protected]
<mailto:[email protected]>.
CSeq: 1 INVITE.
Server: Asterisk PBX UNKNOWN__and_probably_unsupported.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH.
Supported: replaces, timer.
Contact: <sip:[email protected]:5060
<http://sip:[email protected]:5060>>.
Content-Type: application/sdp.
Content-Length: 329.
.
v=0.
o=root 1860889533 1860889534 IN IP4 192.168.2.10.
s=Asterisk PBX UNKNOWN__and_probably_unsupported.
c=IN IP4 192.168.2.5.
t=0 0.
m=audio 31148 RTP/AVP 18 101.
a=rtpmap:18 G729/8000.
a=fmtp:18 annexb=no.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=silenceSupp:off - - - -.
a=ptime:20.
a=sendrecv.
a=nortpproxy:yes.
U 2013/04/09 12:17:06.988918 108.59.2.133:5060
<http://108.59.2.133:5060> -> 192.168.2.5:5060 <http://192.168.2.5:5060>
BYE sip:[email protected]:5060 <http://sip:[email protected]:5060>
SIP/2.0.
Max-Forwards: 64.
To: "1001" <sip:[email protected]
<mailto:sip%[email protected]>>;tag=as4b40d9b4.
From: <sip:[email protected]
<mailto:sip%[email protected]>>;tag=3574513019-870807.
Reason: Q.850;cause=16;text="".
Call-ID: [email protected]:5060
<http://[email protected]:5060>.
CSeq: 2 BYE.
Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO,
REFER, SUBSCRIBE, PRACK, UPDATE.
Via: SIP/2.0/UDP 108.59.2.133;branch=z9hG4bK2deb.8bfd0b06.0.
Contact: <sip:[email protected]
<mailto:sip%[email protected]>;did=e9e.a6618961>.
Allow-Events: as-feature-event.
Allow-Events: call-info.
Allow-Events: presence.
Allow-Events: line-seize.
Allow-Events: dialog.
Allow-Events: refer.
Allow-Events: message-summary.
Content-Length: 0.
.
Forcing RPORT: sip:[email protected]
<mailto:sip%[email protected]>
U 2013/04/09 12:17:06.989421 192.168.2.5:5060
<http://192.168.2.5:5060> -> 108.59.2.133:5060 <http://108.59.2.133:5060>
SIP/2.0 404 Not here.
To: "1001" <sip:[email protected]
<mailto:sip%[email protected]>>;tag=as4b40d9b4.
From: <sip:[email protected]
<mailto:sip%[email protected]>>;tag=3574513019-870807.
Call-ID: [email protected]:5060
<http://[email protected]:5060>.
CSeq: 2 BYE.
Via: SIP/2.0/UDP
108.59.2.133;received=108.59.2.133;rport=5060;branch=z9hG4bK2deb.8bfd0b06.0.
Content-Length: 0.
Or is asterisk the culprit? Looking at the forwarded INVITE (on the
asterisk server), I see that the RR has been re-written, as opposed to
appended when contacting the provider:
U 2013/04/09 12:52:52.109611 192.168.2.10:5060
<http://192.168.2.10:5060> -> 108.59.2.133:5060 <http://108.59.2.133:5060>
INVITE sip:[email protected]
<mailto:sip%[email protected]> SIP/2.0.
Via: SIP/2.0/UDP 70.10.163.44:5060;branch=z9hG4bK75a764b9;rport.
Max-Forwards: 70.
From: "1001" <sip:[email protected]
<mailto:sip%[email protected]>>;tag=as234a7f7d.
To: <sip:[email protected]
<mailto:sip%[email protected]>>.
Contact: <sip:[email protected]:5060 <http://sip:[email protected]:5060>>.
Call-ID: [email protected]:5060
<http://[email protected]:5060>.
CSeq: 102 INVITE.
User-Agent: Asterisk PBX UNKNOWN__and_probably_unsupported.
Date: Tue, 09 Apr 2013 16:52:52 GMT.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH.
Supported: replaces, timer.
Content-Type: application/sdp.
Content-Length: 310.
.
v=0.
o=root 731333659 731333659 IN IP4 70.10.163.44.
s=Asterisk PBX UNKNOWN__and_probably_unsupported.
c=IN IP4 70.10.163.44.
t=0 0.
m=audio 30434 RTP/AVP 18 101.
a=rtpmap:18 G729/8000.
a=fmtp:18 annexb=no.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=silenceSupp:off - - - -.
a=ptime:20.
a=sendrecv.
Can we get an externally initiated BYE working in an
OpenSIPS->Asterisk integration? If so, some suggestions would be
appreciated. Maybe just really the non-loose route BYE to asterisk?
Is adding topology hiding functionality a cumbersome task...
Thanks in Advance,
N.
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