[Serusers] HELP: SER-0.9 Record Route Problem Between SER andAsterisk

Greger V. Teigre greger at teigre.com
Thu Feb 24 08:52:20 CET 2005


Paul,
I'm stretching my RFC understanding here, but I'll give my 2c:
I agree with you that a route header is missing in the Asterisk-generated 
BYE. The question is why.  I can see no problems with your record-routing 
and I doubt that Asterisk does something wrong. The only anomaly I can see 
is the 68.86.100.20 Record-Route:
Record-Route: <sip:68.86.100.20:5060;lr>.

The way I read the RFC, lr should be specified as lr=on if the proxy 
supports loose routing.  May it be that Asterisk interprets the proxy as a 
strict router and thus (correctly) does not include a Route header?  From 
the RFC, section 19.1.1:

      URI parameters: Parameters affecting a request constructed from
         the URI. URI parameters are added after the hostport component and 
are
         separated by semi-colons.

         URI parameters take the form:
            parameter-name "=" parameter-value

         Even though an arbitrary number of URI parameters may be
         included in a URI, any given parameter-name MUST NOT appear
         more than once. This extensible mechanism includes the transport, 
maddr, ttl,
         user, method and lr parameters.

g-)

Java Rockx wrote:
> Hi All.
>
> I'm using ser-0.9
>
> Can anyone take a quick look at this short SIP conversation and tell
> me if they think the problem is with my ser.cfg or a bug in Asterisk
> 1.0.2.
>
> We use a 3rd party for PSTN gateway functionality. This 3rd party uses
> a Sonus box behind a SIP proxy. Our SER proxy talks directly to their
> SIP proxy as needed to complete PSTN calls.
>
> The problem is that when a PSTN caller dials a SIP phone and gets sent
> to voice mail (Asterisk) because of a no answer or busy condition,
> Asterisk hangs up after the caller leaves a message. When Asterisk
> hangs up, the BYE from Asterisk is sent to SER, however, SER
> incorrectly forwards the BYE directly to their Sonus gateway, rather
> than the their SIP proxy. This causes our PSTN gateway provider to
> have "open" billing records in their system.
>
> If you look at the BYE message from Asterisk to SER you can see that
> route headers are missing (I think). The final BYE should have been
> sent to 68.86.100.20, but it was sent to 68.86.100.30 instead.
>
> I am record_route()ing all messages except for REGISTER and I have the
> mhomed=1 parameter set.
>
> Can anyone help me put the blame on either my ser.cfg or Asterisk?
>
> Regards,
> Paul
>
>
> IP LEGEND
> -----------
> 68.86.100.30 - 3rd Party Sonus PSTN Gateway
> 68.86.100.20 - 3rd Party SIP Proxy
> 24.11.12.24  - Sip Express Router (eth0)
> 10.255.255.1 - Sip Express Router (eth1)
> 10.255.255.2 - Asterisk PBX
>
> NOTE: I have Asterisk connected to the SER server with a crossover
> cable.
>
>
>
> U 2005/02/23 22:24:18.848582 68.86.100.20:5060 -> 24.11.12.24:5060
> INVITE sip:4075551212 at 24.11.12.24:5060 SIP/2.0.
> Via: SIP/2.0/UDP 68.86.100.20:5060;branch=z9hG4bKed82c1ba766-c3014f40.
> Via: SIP/2.0/UDP
> 68.86.100.30:5060;branch=7a874abcac87c7752a3d3c5c7ba10dc0.
> To: 4075551212 <sip:4075551212 at 68.86.100.30:5060>.
> From: sip:3211231234 at 66.236.245.98;tag=27DECB5C-17BD.
> Call-ID: 9028535-3318204258-749010 at 68.86.100.30.
> CSeq: 1 INVITE.
> Max-Forwards: 4.
> Contact: sip:3211231234 at 68.86.100.30:5060.
> Record-Route: <sip:68.86.100.20:5060;lr>.
> Content-Type: application/sdp.
> Content-Length: 312.
> .
> v=0.
> o=NexTone-MSW 1234 187 IN IP4 66.236.245.98.
> s=sip call.
> c=IN IP4 66.236.245.98.
> t=0 0.
> m=audio 16814 RTP/AVP 18 0 4 8 101.
> a=rtpmap:18 G729/8000.
> a=fmtp:18 annexb=no.
> a=rtpmap:0 PCMU/8000.
> a=rtpmap:4 G723/8000.
> a=fmtp:4 annexa=yes.
> a=rtpmap:8 PCMA/8000.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-16.
>
> #
> U 2005/02/23 22:24:18.860022 24.11.12.24:5060 -> 68.86.100.20:5060
> SIP/2.0 100 trying -- your call is important to us.
> Via: SIP/2.0/UDP 68.86.100.20:5060;branch=z9hG4bKed82c1ba766-c3014f40.
> Via: SIP/2.0/UDP
> 68.86.100.30:5060;branch=7a874abcac87c7752a3d3c5c7ba10dc0.
> To: 4075551212 <sip:4075551212 at 68.86.100.30:5060>.
> From: sip:3211231234 at 66.236.245.98;tag=27DECB5C-17BD.
> Call-ID: 9028535-3318204258-749010 at 68.86.100.30.
> CSeq: 1 INVITE.
> Content-Length: 0.
> .
>
> #
> U 2005/02/23 22:24:18.860259 10.255.255.1:1033 -> 10.255.255.2:5060
> INVITE sip:699 at 10.255.255.2:5060 SIP/2.0.
> Record-Route: <sip:10.255.255.1;r2=on;ftag=27DECB5C-17BD;lr=on>.
> Record-Route: <sip:24.11.12.24;r2=on;ftag=27DECB5C-17BD;lr=on>.
> Via: SIP/2.0/UDP 10.255.255.1;branch=z9hG4bKb929.21080974.0.
> Via: SIP/2.0/UDP 68.86.100.20:5060;branch=z9hG4bKed82c1ba766-c3014f40.
> Via: SIP/2.0/UDP
> 68.86.100.30:5060;branch=7a874abcac87c7752a3d3c5c7ba10dc0.
> To: 4075551212 <sip:4075551212 at 68.86.100.30:5060>.
> From: sip:3211231234 at 66.236.245.98;tag=27DECB5C-17BD.
> Call-ID: 9028535-3318204258-749010 at 68.86.100.30.
> CSeq: 1 INVITE.
> Max-Forwards: 3.
> Contact: sip:3211231234 at 68.86.100.30:5060.
> Record-Route: <sip:68.86.100.20:5060;lr>.
> Content-Type: application/sdp.
> Content-Length: 312.
> .
> v=0.
> o=NexTone-MSW 1234 187 IN IP4 66.236.245.98.
> s=sip call.
> c=IN IP4 24.11.12.24.
> t=0 0.
> m=audio 36574 RTP/AVP 18 0 4 8 101.
> a=rtpmap:18 G729/8000.
> a=fmtp:18 annexb=no.
> a=rtpmap:0 PCMU/8000.
> a=rtpmap:4 G723/8000.
> a=fmtp:4 annexa=yes.
> a=rtpmap:8 PCMA/8000.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-16.
>
> #
> U 2005/02/23 22:24:18.871131 10.255.255.2:5060 -> 10.255.255.1:1033
> SIP/2.0 100 Trying.
> Via: SIP/2.0/UDP
> 10.255.255.1;branch=z9hG4bKb929.21080974.0;received=10.255.255.1;rport=1033.
> Via: SIP/2.0/UDP 68.86.100.20:5060;branch=z9hG4bKed82c1ba766-c3014f40.
> Via: SIP/2.0/UDP
> 68.86.100.30:5060;branch=7a874abcac87c7752a3d3c5c7ba10dc0.
> From: sip:3211231234 at 66.236.245.98;tag=27DECB5C-17BD.
> To: 4075551212 <sip:4075551212 at 68.86.100.30:5060>;tag=as588114d9.
> Call-ID: 9028535-3318204258-749010 at 68.86.100.30.
> CSeq: 1 INVITE.
> User-Agent: Asterisk PBX.
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER.
> Contact: <sip:699 at 10.255.255.2>.
> Content-Length: 0.
> .
>
> #
> U 2005/02/23 22:24:18.879160 10.255.255.2:5060 -> 10.255.255.1:1033
> SIP/2.0 200 OK.
> Via: SIP/2.0/UDP
> 10.255.255.1;branch=z9hG4bKb929.21080974.0;received=10.255.255.1;rport=1033.
> Via: SIP/2.0/UDP 68.86.100.20:5060;branch=z9hG4bKed82c1ba766-c3014f40.
> Via: SIP/2.0/UDP
> 68.86.100.30:5060;branch=7a874abcac87c7752a3d3c5c7ba10dc0.
> Record-Route: <sip:10.255.255.1;r2=on;ftag=27DECB5C-17BD;lr=on>.
> Record-Route: <sip:24.11.12.24;r2=on;ftag=27DECB5C-17BD;lr=on>.
> Record-Route: <sip:68.86.100.20:5060;lr>.
> From: sip:3211231234 at 66.236.245.98;tag=27DECB5C-17BD.
> To: 4075551212 <sip:4075551212 at 68.86.100.30:5060>;tag=as588114d9.
> Call-ID: 9028535-3318204258-749010 at 68.86.100.30.
> CSeq: 1 INVITE.
> User-Agent: Asterisk PBX.
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER.
> Contact: <sip:699 at 10.255.255.2>.
> Content-Type: application/sdp.
> Content-Length: 362.
> .
> v=0.
> o=root 550 550 IN IP4 10.255.255.2.
> s=session.
> c=IN IP4 10.255.255.2.
> t=0 0.
> m=audio 17900 RTP/AVP 97 18 3 4 2 0 8 101.
> a=rtpmap:97 iLBC/8000.
> a=rtpmap:18 G729/8000.
> a=rtpmap:3 GSM/8000.
> a=rtpmap:4 G723/8000.
> a=rtpmap:2 G726-32/8000.
> a=rtpmap:0 PCMU/8000.
> a=rtpmap:8 PCMA/8000.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-16.
> a=silenceSupp:off - - - -.
>
> #
> U 2005/02/23 22:24:18.883882 24.11.12.24:5060 -> 68.86.100.20:5060
> SIP/2.0 200 OK.
> Via: SIP/2.0/UDP 68.86.100.20:5060;branch=z9hG4bKed82c1ba766-c3014f40.
> Via: SIP/2.0/UDP
> 68.86.100.30:5060;branch=7a874abcac87c7752a3d3c5c7ba10dc0.
> Record-Route: <sip:10.255.255.1;r2=on;ftag=27DECB5C-17BD;lr=on>.
> Record-Route: <sip:24.11.12.24;r2=on;ftag=27DECB5C-17BD;lr=on>.
> Record-Route: <sip:68.86.100.20:5060;lr>.
> From: sip:3211231234 at 66.236.245.98;tag=27DECB5C-17BD.
> To: 4075551212 <sip:4075551212 at 68.86.100.30:5060>;tag=as588114d9.
> Call-ID: 9028535-3318204258-749010 at 68.86.100.30.
> CSeq: 1 INVITE.
> User-Agent: Asterisk PBX.
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER.
> Contact: <sip:699 at 10.255.255.2>.
> Content-Type: application/sdp.
> Content-Length: 363.
> .
> v=0.
> o=root 550 550 IN IP4 10.255.255.2.
> s=session.
> c=IN IP4 24.11.12.24.
> t=0 0.
> m=audio 36574 RTP/AVP 97 18 3 4 2 0 8 101.
> a=rtpmap:97 iLBC/8000.
> a=rtpmap:18 G729/8000.
> a=rtpmap:3 GSM/8000.
> a=rtpmap:4 G723/8000.
> a=rtpmap:2 G726-32/8000.
> a=rtpmap:0 PCMU/8000.
> a=rtpmap:8 PCMA/8000.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-16.
> a=silenceSupp:off - - - -.
>
> #
> U 2005/02/23 22:24:19.097436 68.86.100.20:5060 -> 24.11.12.24:5060
> ACK sip:699 at 10.255.255.2 SIP/2.0.
> Via: SIP/2.0/UDP
> 68.86.100.20:5060;branch=ef5ce1f8a400adab242436d0564cb045.
> Via: SIP/2.0/UDP
> 68.86.100.30:5060;branch=ef5ce1f8a400adab242436d0564cb045.
> To: 4075551212 <sip:4075551212 at 68.86.100.30:5060>;tag=as588114d9.
> From: sip:3211231234 at 66.236.245.98;tag=27DECB5C-17BD.
> Call-ID: 9028535-3318204258-749010 at 68.86.100.30.
> CSeq: 1 ACK.
> Max-Forwards: 4.
> Contact: sip:3211231234 at 68.86.100.30:5060.
> Record-Route: <sip:68.86.100.20:5060;lr>.
> Route: <sip:24.11.12.24;r2=on;ftag=27DECB5C-17BD;lr=on>.
> Route: <sip:10.255.255.1;r2=on;ftag=27DECB5C-17BD;lr=on>.
> Content-Length: 0.
> .
>
> #
> U 2005/02/23 22:24:19.098087 10.255.255.1:1033 -> 10.255.255.2:5060
> ACK sip:699 at 10.255.255.2 SIP/2.0.
> Record-Route: <sip:10.255.255.1;r2=on;ftag=27DECB5C-17BD;lr=on>.
> Record-Route: <sip:24.11.12.24;r2=on;ftag=27DECB5C-17BD;lr=on>.
> Via: SIP/2.0/UDP 10.255.255.1;branch=0.
> Via: SIP/2.0/UDP
> 68.86.100.20:5060;branch=ef5ce1f8a400adab242436d0564cb045.
> Via: SIP/2.0/UDP
> 68.86.100.30:5060;branch=ef5ce1f8a400adab242436d0564cb045.
> To: 4075551212 <sip:4075551212 at 68.86.100.30:5060>;tag=as588114d9.
> From: sip:3211231234 at 66.236.245.98;tag=27DECB5C-17BD.
> Call-ID: 9028535-3318204258-749010 at 68.86.100.30.
> CSeq: 1 ACK.
> Max-Forwards: 3.
> Contact: sip:3211231234 at 68.86.100.30:5060.
> Record-Route: <sip:68.86.100.20:5060;lr>.
> Content-Length: 0.
> .
>
> ###
> U 2005/02/23 22:24:25.104860 10.255.255.2:5060 -> 10.255.255.1:1033
> BYE sip:3211231234 at 68.86.100.30:5060 SIP/2.0.
> Via: SIP/2.0/UDP 10.255.255.2:5060;branch=z9hG4bK2da77693;rport.
> Route:
> <sip:24.11.12.24;r2=on;ftag=27DECB5C-17BD;lr=on>,<sip:68.86.100.20:5060;lr>,<sip:3211231234 at 68.86.100.30:5060>.
> From: 4075551212 <sip:4075551212 at 68.86.100.30:5060>;tag=as588114d9.
> To: sip:3211231234 at 66.236.245.98;tag=27DECB5C-17BD.
> Contact: <sip:699 at 10.255.255.2>.
> Call-ID: 9028535-3318204258-749010 at 68.86.100.30.
> CSeq: 102 BYE.
> User-Agent: Asterisk PBX.
> Content-Length: 0.
> .
>
> #
> U 2005/02/23 22:24:25.108961 24.11.12.24:5060 -> 68.86.100.30:5060
> BYE sip:3211231234 at 68.86.100.30:5060 SIP/2.0.
> Max-Forwards: 10.
> Record-Route: <sip:24.11.12.24;r2=on;ftag=as588114d9;lr=on>.
> Record-Route: <sip:10.255.255.1;r2=on;ftag=as588114d9;lr=on>.
> Via: SIP/2.0/UDP 24.11.12.24;branch=z9hG4bK231b.07949ea5.0.
> Via: SIP/2.0/UDP 10.255.255.2:5060;branch=z9hG4bK2da77693;rport=5060.
> From: 4075551212 <sip:4075551212 at 68.86.100.30:5060>;tag=as588114d9.
> To: sip:3211231234 at 66.236.245.98;tag=27DECB5C-17BD.
> Contact: <sip:699 at 10.255.255.2>.
> Call-ID: 9028535-3318204258-749010 at 68.86.100.30.
> CSeq: 102 BYE.
> User-Agent: Asterisk PBX.
> Content-Length: 0.
> Route: <sip:3211231234 at 68.86.100.30:5060>.
> .
>
> #
> U 2005/02/23 22:24:25.175832 68.86.100.30:5060 -> 24.11.12.24:5060
> SIP/2.0 200 OK.
> Via: SIP/2.0/UDP 24.11.12.24;branch=z9hG4bK231b.07949ea5.0.
> Via: SIP/2.0/UDP 10.255.255.2:5060;branch=z9hG4bK2da77693;rport=5060.
> From: 4075551212 <sip:4075551212 at 68.86.100.30:5060>;tag=as588114d9.
> To: sip:3211231234 at 66.236.245.98;tag=27DECB5C-17BD.
> Call-ID: 9028535-3318204258-749010 at 68.86.100.30.
> CSeq: 102 BYE.
> Content-Length: 0.
> .
>
> #
> U 2005/02/23 22:24:25.176182 10.255.255.1:1033 -> 10.255.255.2:5060
> SIP/2.0 200 OK.
> Via: SIP/2.0/UDP 10.255.255.2:5060;branch=z9hG4bK2da77693;rport=5060.
> From: 4075551212 <sip:4075551212 at 68.86.100.30:5060>;tag=as588114d9.
> To: sip:3211231234 at 66.236.245.98;tag=27DECB5C-17BD.
> Call-ID: 9028535-3318204258-749010 at 68.86.100.30.
> CSeq: 102 BYE.
> Content-Length: 0.
> .
>
> _______________________________________________
> Serusers mailing list
> serusers at lists.iptel.org
> http://lists.iptel.org/mailman/listinfo/serusers 




More information about the sr-users mailing list