Re: [OpenSIPS-Users] 404 When BYE initiated by external callee

Wed, 10 Apr 2013 01:51:48 -0700 

Nick,
I do not know what is the topology of your SIP network, but the idea is that the BYE received by OpenSIPS does not contain proper routing information - now, either the BYE was wrongly generated by the end point, either it was wrongly changed on the way (if there are more hops between that end point and opensips). 

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com


On 04/09/2013 09:23 PM, Nick Khamis wrote:

On Tue, Apr 9, 2013 at 1:22 PM, Bogdan-Andrei Iancu 
<[email protected] <mailto:[email protected]>> wrote:


    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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Is our asterisk server not relaying the RR along with the INVITE? If 
so, can we configure the PBX to do so using one of it's variables? * 
Mailing list CC'ed in this email...



N.

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