Hi,
We are having issues where the "OK" or "ACK" is that is coming from the phone is not relayed by OpenSER to Asterisk.
Below is the sip trace... I am also attaching a tcpdump. Please help what we can do.
Received from udp:10.1.10.80:5060 at 26/10/2011 10:22:41:476 (490 bytes):
SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/UDP 10.30.0.64:5060 ;received=10.30.0.64;branch=z9hG4bK-wowp1kmdy4rl;rport=5060 From: "Virgil Menendez" sip:91421@ser.gowireless.net;tag=6wkdms1r20 To: sip:9513261429@ser.gowireless.net;user=phone;tag=as0b87218f Call-ID: 3c26755bf15c-9iq08xqqblo6 CSeq: 4 INVITE Server: Asterisk PBX 1.8.7.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0
------------------------------
Sent to udp:10.1.10.80:5060 at 26/10/2011 10:22:41:481 (387 bytes):
ACK sip:vm9513261429@10.1.10.83:5060 SIP/2.0 v: SIP/2.0/UDP 10.30.0.64:5060;branch=z9hG4bK-wowp1kmdy4rl;rport Route: sip:10.1.10.80;lr=on f: "Virgil Menendez" sip:91421@ser.gowireless.net;tag=6wkdms1r20 t: sip:9513261429@ser.gowireless.net;user=phone;tag=as0b87218f i: 3c26755bf15c-9iq08xqqblo6 CSeq: 4 ACK Max-Forwards: 70 m: sip:91421@10.30.0.64:5060;reg-id=1 l: 0
------------------------------
Received from udp:10.1.10.80:5060 at 26/10/2011 10:22:42:130 (868 bytes):
SIP/2.0 200 OK Via: SIP/2.0/UDP 10.30.0.64:5060 ;received=10.30.0.64;branch=z9hG4bK-5evtiw6dm0po;rport=5060 Record-Route: sip:10.1.10.80;lr=on From: "Virgil Menendez" sip:91421@ser.gowireless.net;tag=qi3i8ze6z8 To: sip:9513261429@ser.gowireless.net;user=phone;tag=as3f8c0f96 Call-ID: 3c2676547a8d-2t5yi6jok1sv CSeq: 2 INVITE Server: Asterisk PBX 1.8.7.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: sip:9513261429@10.1.10.83:5060 Content-Type: application/sdp Content-Length: 256
v=0 o=root 1355451627 1355451627 IN IP4 10.1.10.83 s=Asterisk PBX 1.8.7.1 c=IN IP4 10.1.10.83 t=0 0 m=audio 16094 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv ------------------------------
Sent to udp:10.1.10.80:5060 at 26/10/2011 10:22:42:132 (385 bytes):
ACK sip:9513261429@10.1.10.83:5060 SIP/2.0 v: SIP/2.0/UDP 10.30.0.64:5060;branch=z9hG4bK-wszafb7cbzpw;rport Route: sip:10.1.10.80;lr=on f: "Virgil Menendez" sip:91421@ser.gowireless.net;tag=qi3i8ze6z8 t: sip:9513261429@ser.gowireless.net;user=phone;tag=as3f8c0f96 i: 3c2676547a8d-2t5yi6jok1sv CSeq: 2 ACK Max-Forwards: 70 m: sip:91421@10.30.0.64:5060;reg-id=1 l: 0
------------------------------
Received from udp:10.1.10.80:5060 at 26/10/2011 10:22:42:232 (503 bytes):
BYE sip:91421@10.30.0.64:5060 SIP/2.0 Via: SIP/2.0/UDP 10.1.10.80;branch=z9hG4bKe723.bf70c1f4.0 Via: SIP/2.0/UDP 10.1.10.83:5060;branch=z9hG4bK69f53cf1 Max-Forwards: 69 From: sip:9513261429@ser.gowireless.net;user=phone;tag=as3f8c0f96 To: "Virgil Menendez" sip:91421@ser.gowireless.net;tag=qi3i8ze6z8 Call-ID: 3c2676547a8d-2t5yi6jok1sv CSeq: 102 BYE User-Agent: Asterisk PBX 1.8.7.1 *X-Asterisk-HangupCause: Protocol error, unspecified *X-Asterisk-HangupCauseCode: 111 Content-Length: 0
Regards,
Rowell
Hello,
you have to provide the sip trace taken on the sip server, in order to see what is received and what is sent out by kamailio. Looks like the one you pasted here is from client.
You can use ngrep on kamailio server:
ngrep -d any -qt -W byline port 5060
Also, the packets you pasted next are from two different calls (see the call-id header). The second seems to be completed ok, but something is not good for asterisk and it sends bye. Maybe you can spot something in the logs of asterisk.
Cheers, Daniel
On 10/26/11 8:25 PM, Rowell Rufino wrote:
Hi, We are having issues where the "OK" or "ACK" is that is coming from the phone is not relayed by OpenSER to Asterisk. Below is the sip trace... I am also attaching a tcpdump. Please help what we can do.
Received from udp:10.1.10.80:5060 http://10.1.10.80:5060 at 26/10/2011 10:22:41:476 (490 bytes):
SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/UDP 10.30.0.64:5060;received=10.30.0.64;branch=z9hG4bK-wowp1kmdy4rl;rport=5060 From: "Virgil Menendez" <sip:91421@ser.gowireless.net mailto:sip%3A91421@ser.gowireless.net>;tag=6wkdms1r20 To: <sip:9513261429@ser.gowireless.net mailto:sip%3A9513261429@ser.gowireless.net;user=phone>;tag=as0b87218f Call-ID: 3c26755bf15c-9iq08xqqblo6 CSeq: 4 INVITE Server: Asterisk PBX 1.8.7.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0
Sent to udp:10.1.10.80:5060 http://10.1.10.80:5060 at 26/10/2011 10:22:41:481 (387 bytes):
ACK sip:vm9513261429@10.1.10.83:5060 http://sip:vm9513261429@10.1.10.83:5060 SIP/2.0 v: SIP/2.0/UDP 10.30.0.64:5060;branch=z9hG4bK-wowp1kmdy4rl;rport Route: sip:10.1.10.80;lr=on f: "Virgil Menendez" <sip:91421@ser.gowireless.net mailto:sip%3A91421@ser.gowireless.net>;tag=6wkdms1r20 t: <sip:9513261429@ser.gowireless.net mailto:sip%3A9513261429@ser.gowireless.net;user=phone>;tag=as0b87218f i: 3c26755bf15c-9iq08xqqblo6 CSeq: 4 ACK Max-Forwards: 70 m: <sip:91421@10.30.0.64:5060 http://sip:91421@10.30.0.64:5060>;reg-id=1 l: 0
Received from udp:10.1.10.80:5060 http://10.1.10.80:5060 at 26/10/2011 10:22:42:130 (868 bytes):
SIP/2.0 200 OK Via: SIP/2.0/UDP 10.30.0.64:5060;received=10.30.0.64;branch=z9hG4bK-5evtiw6dm0po;rport=5060 Record-Route: sip:10.1.10.80;lr=on From: "Virgil Menendez" <sip:91421@ser.gowireless.net mailto:sip%3A91421@ser.gowireless.net>;tag=qi3i8ze6z8 To: <sip:9513261429@ser.gowireless.net mailto:sip%3A9513261429@ser.gowireless.net;user=phone>;tag=as3f8c0f96 Call-ID: 3c2676547a8d-2t5yi6jok1sv CSeq: 2 INVITE Server: Asterisk PBX 1.8.7.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: <sip:9513261429@10.1.10.83:5060 http://sip:9513261429@10.1.10.83:5060> Content-Type: application/sdp Content-Length: 256
v=0 o=root 1355451627 1355451627 IN IP4 10.1.10.83 s=Asterisk PBX 1.8.7.1 c=IN IP4 10.1.10.83 t=0 0 m=audio 16094 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv
Sent to udp:10.1.10.80:5060 http://10.1.10.80:5060 at 26/10/2011 10:22:42:132 (385 bytes):
ACK sip:9513261429@10.1.10.83:5060 http://sip:9513261429@10.1.10.83:5060 SIP/2.0 v: SIP/2.0/UDP 10.30.0.64:5060;branch=z9hG4bK-wszafb7cbzpw;rport Route: sip:10.1.10.80;lr=on f: "Virgil Menendez" <sip:91421@ser.gowireless.net mailto:sip%3A91421@ser.gowireless.net>;tag=qi3i8ze6z8 t: <sip:9513261429@ser.gowireless.net mailto:sip%3A9513261429@ser.gowireless.net;user=phone>;tag=as3f8c0f96 i: 3c2676547a8d-2t5yi6jok1sv CSeq: 2 ACK Max-Forwards: 70 m: <sip:91421@10.30.0.64:5060 http://sip:91421@10.30.0.64:5060>;reg-id=1 l: 0
Received from udp:10.1.10.80:5060 http://10.1.10.80:5060 at 26/10/2011 10:22:42:232 (503 bytes):
BYE sip:91421@10.30.0.64:5060 http://sip:91421@10.30.0.64:5060 SIP/2.0 Via: SIP/2.0/UDP 10.1.10.80;branch=z9hG4bKe723.bf70c1f4.0 Via: SIP/2.0/UDP 10.1.10.83:5060;branch=z9hG4bK69f53cf1 Max-Forwards: 69 From: <sip:9513261429@ser.gowireless.net mailto:sip%3A9513261429@ser.gowireless.net;user=phone>;tag=as3f8c0f96 To: "Virgil Menendez" <sip:91421@ser.gowireless.net mailto:sip%3A91421@ser.gowireless.net>;tag=qi3i8ze6z8 Call-ID: 3c2676547a8d-2t5yi6jok1sv CSeq: 102 BYE User-Agent: Asterisk PBX 1.8.7.1 *X-Asterisk-HangupCause: Protocol error, unspecified *X-Asterisk-HangupCauseCode: 111 Content-Length: 0
Regards,
Rowell
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-
Daniel-Constantin Mierla -
Rowell Rufino