I recently installed the latest version of openser and this time used mediaproxy rather than rtpproxy. Everything seems to work but if a sip device is called the phone rings and is instantally disconnected and the far end is left off-hook. This worked before but I did modify my script to work with mediaproxy. Below is the wireshark decode of the sip messagining. Any help or suggestions on where to look would be great. Calls from the sip device works flawlessly. I am using a MaxTNT as the gateway.
|Time | 152.93.36.91 | siprt1.me.net| 152.93.37.83 |
|22.031 | INVITE SDP ( telephone-event) | |SIP From: sip:8385021101@152.53.16.91:5060 To:sip: 8385024200@ siprt1.me.net:5060
| |(5060) ------------------> (5060) | |
|22.031 | 100 Giving a try | |SIP Status
| |(5060) <------------------ (5060) | |
|22.031 | | INVITE SDP ( telephone-event) |SIP Request
| | |(5060) ------------------> (5060) |
|22.040 | | 100 Trying| |SIP Status
| | |(5060) <------------------ (5060) |
|22.042 | | 180 Ringing |SIP Status
| | |(5060) <------------------ (5060) |
|22.042 | 180 Ringing | |SIP Status
| |(5060) <------------------ (5060) | |
|25.244 | | 200 OK SDP ( telephone-event) |SIP Status
| | |(5060) <------------------ (5060) |
|25.245 | 200 OK SDP ( telephone-event) | |SIP Status
| |(5060) <------------------ (5060) | |
|25.269 | ACK | | |SIP Request
| |(5060) ------------------> (5060) | |
|25.269 | | ACK | |SIP Request
| | |(5060) ------------------> (5060) |
|25.269 | INVITE SDP ( telephone-event) | |SIP From: sip: 8385021101@152.93.36.91:5060 To:sip: 8385024200@ siprt1.me.net:5060
| |(5060) ------------------> (5060) | |
|25.270 | 407 Proxy Authentication Required | |SIP Status
| |(5060) <------------------ (5060) | |
|25.291 | ACK | | |SIP Request
| |(5060) ------------------> (5060) | |
|25.291 | BYE | | |SIP Request
| |(5060) ------------------> (5060) | |
|25.293 | | BYE | |SIP Request
| | |(5060) ------------------> (5060) |
|25.326 | | 200 OK | |SIP Status
| | |(5060) <------------------ (5060) |
|25.327 | 200 OK | | |SIP Status
| |(5060) <------------------ (5060) | |
Shane
Maybe a bug in the caller's SIP client?
regards klaus
Shane Burrell wrote:
I recently installed the latest version of openser and this time used mediaproxy rather than rtpproxy. Everything seems to work but if a sip device is called the phone rings and is instantally disconnected and the far end is left off-hook. This worked before but I did modify my script to work with mediaproxy. Below is the wireshark decode of the sip messagining. Any help or suggestions on where to look would be great. Calls from the sip device works flawlessly. I am using a MaxTNT as the gateway.
|Time | 152.93.36.91 | siprt1.me.net| 152.93.37.83 |
|22.031 | INVITE SDP ( telephone-event) | |SIP From: sip:8385021101@152.53.16.91:5060 To:sip: 8385024200@ siprt1.me.net:5060
| |(5060) ------------------> (5060) | |
|22.031 | 100 Giving a try | |SIP Status
| |(5060) <------------------ (5060) | |
|22.031 | | INVITE SDP ( telephone-event) |SIP Request
| | |(5060) ------------------> (5060) |
|22.040 | | 100 Trying| |SIP Status
| | |(5060) <------------------ (5060) |
|22.042 | | 180 Ringing |SIP Status
| | |(5060) <------------------ (5060) |
|22.042 | 180 Ringing | |SIP Status
| |(5060) <------------------ (5060) | |
|25.244 | | 200 OK SDP ( telephone-event) |SIP Status
| | |(5060) <------------------ (5060) |
|25.245 | 200 OK SDP ( telephone-event) | |SIP Status
| |(5060) <------------------ (5060) | |
|25.269 | ACK | | |SIP Request
| |(5060) ------------------> (5060) | |
|25.269 | | ACK | |SIP Request
| | |(5060) ------------------> (5060) |
|25.269 | INVITE SDP ( telephone-event) | |SIP From: sip: 8385021101@152.93.36.91:5060 To:sip: 8385024200@ siprt1.me.net:5060
| |(5060) ------------------> (5060) | |
|25.270 | 407 Proxy Authentication Required | |SIP Status
| |(5060) <------------------ (5060) | |
|25.291 | ACK | | |SIP Request
| |(5060) ------------------> (5060) | |
|25.291 | BYE | | |SIP Request
| |(5060) ------------------> (5060) | |
|25.293 | | BYE | |SIP Request
| | |(5060) ------------------> (5060) |
|25.326 | | 200 OK | |SIP Status
| | |(5060) <------------------ (5060) |
|25.327 | 200 OK | | |SIP Status
| |(5060) <------------------ (5060) | |
Shane
Users mailing list Users@openser.org http://openser.org/cgi-bin/mailman/listinfo/users
The SIP UA was a grandstream ATA running the latest stable firmware. Prior to upgrading to 1.1 and moving to mediaproxy it worked well with the exception of good nat support which is why I would really like mediaproxy to work. Is there anything I should look for in the sip dialog to determine if the client, sip proxy, or the gateway is the culprit on disconnecting the call?
Thanks,
Shane
-----Original Message----- From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] Sent: Wednesday, January 03, 2007 4:52 AM To: Shane Burrell Cc: users@openser.org Subject: Re: [Users] Issues with calls using openser.
Maybe a bug in the caller's SIP client?
regards klaus
Shane Burrell wrote:
I recently installed the latest version of openser and this time used mediaproxy rather than rtpproxy. Everything seems to work but if a sip device is called the phone rings and is instantally disconnected and the
far
end is left off-hook. This worked before but I did modify my script to
work
with mediaproxy. Below is the wireshark decode of the sip messagining.
Any
help or suggestions on where to look would be great. Calls from the sip device works flawlessly. I am using a MaxTNT as the gateway.
|Time | 152.93.36.91 | siprt1.me.net| 152.93.37.83 |
|22.031 | INVITE SDP ( telephone-event) | |SIP From: sip:8385021101@152.53.16.91:5060 To:sip: 8385024200@ siprt1.me.net:5060
| |(5060) ------------------> (5060) | |
|22.031 | 100 Giving a try | |SIP Status
| |(5060) <------------------ (5060) | |
|22.031 | | INVITE SDP ( telephone-event) |SIP Request
| | |(5060) ------------------> (5060) |
|22.040 | | 100 Trying| |SIP Status
| | |(5060) <------------------ (5060) |
|22.042 | | 180 Ringing |SIP Status
| | |(5060) <------------------ (5060) |
|22.042 | 180 Ringing | |SIP Status
| |(5060) <------------------ (5060) | |
|25.244 | | 200 OK SDP ( telephone-event) |SIP Status
| | |(5060) <------------------ (5060) |
|25.245 | 200 OK SDP ( telephone-event) | |SIP Status
| |(5060) <------------------ (5060) | |
|25.269 | ACK | | |SIP Request
| |(5060) ------------------> (5060) | |
|25.269 | | ACK | |SIP Request
| | |(5060) ------------------> (5060) |
|25.269 | INVITE SDP ( telephone-event) | |SIP From: sip: 8385021101@152.93.36.91:5060 To:sip: 8385024200@ siprt1.me.net:5060
| |(5060) ------------------> (5060) | |
|25.270 | 407 Proxy Authentication Required | |SIP Status
| |(5060) <------------------ (5060) | |
|25.291 | ACK | | |SIP Request
| |(5060) ------------------> (5060) | |
|25.291 | BYE | | |SIP Request
| |(5060) ------------------> (5060) | |
|25.293 | | BYE | |SIP Request
| | |(5060) ------------------> (5060) |
|25.326 | | 200 OK | |SIP Status
| | |(5060) <------------------ (5060) |
|25.327 | 200 OK | | |SIP Status
| |(5060) <------------------ (5060) | |
Shane
Users mailing list Users@openser.org http://openser.org/cgi-bin/mailman/listinfo/users
The log shows that you challenge reINVITEs. MAybe this breaks the grandstream. Please try without challenging the reINVITE. If this helps, then it is a probably a grandstream bug.
But of course there is the question why the grandstream sends a reINVITE at all? This is often a codec problem.
Please also try to use other client (e.g. xlite) and post complete ngrep dumps: "ngrep -q -t -W byline port 5060"
regards klaus
Shane Burrell wrote:
The SIP UA was a grandstream ATA running the latest stable firmware. Prior to upgrading to 1.1 and moving to mediaproxy it worked well with the exception of good nat support which is why I would really like mediaproxy to work. Is there anything I should look for in the sip dialog to determine if the client, sip proxy, or the gateway is the culprit on disconnecting the call?
Thanks,
Shane
-----Original Message----- From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] Sent: Wednesday, January 03, 2007 4:52 AM To: Shane Burrell Cc: users@openser.org Subject: Re: [Users] Issues with calls using openser.
Maybe a bug in the caller's SIP client?
regards klaus
Shane Burrell wrote:
I recently installed the latest version of openser and this time used mediaproxy rather than rtpproxy. Everything seems to work but if a sip device is called the phone rings and is instantally disconnected and the
far
end is left off-hook. This worked before but I did modify my script to
work
with mediaproxy. Below is the wireshark decode of the sip messagining.
Any
help or suggestions on where to look would be great. Calls from the sip device works flawlessly. I am using a MaxTNT as the gateway.
|Time | 152.93.36.91 | siprt1.me.net| 152.93.37.83 |
|22.031 | INVITE SDP ( telephone-event) | |SIP From: sip:8385021101@152.53.16.91:5060 To:sip: 8385024200@ siprt1.me.net:5060
| |(5060) ------------------> (5060) | |
|22.031 | 100 Giving a try | |SIP Status
| |(5060) <------------------ (5060) | |
|22.031 | | INVITE SDP ( telephone-event) |SIP Request
| | |(5060) ------------------> (5060) |
|22.040 | | 100 Trying| |SIP Status
| | |(5060) <------------------ (5060) |
|22.042 | | 180 Ringing |SIP Status
| | |(5060) <------------------ (5060) |
|22.042 | 180 Ringing | |SIP Status
| |(5060) <------------------ (5060) | |
|25.244 | | 200 OK SDP ( telephone-event) |SIP Status
| | |(5060) <------------------ (5060) |
|25.245 | 200 OK SDP ( telephone-event) | |SIP Status
| |(5060) <------------------ (5060) | |
|25.269 | ACK | | |SIP Request
| |(5060) ------------------> (5060) | |
|25.269 | | ACK | |SIP Request
| | |(5060) ------------------> (5060) |
|25.269 | INVITE SDP ( telephone-event) | |SIP From: sip: 8385021101@152.93.36.91:5060 To:sip: 8385024200@ siprt1.me.net:5060
| |(5060) ------------------> (5060) | |
|25.270 | 407 Proxy Authentication Required | |SIP Status
| |(5060) <------------------ (5060) | |
|25.291 | ACK | | |SIP Request
| |(5060) ------------------> (5060) | |
|25.291 | BYE | | |SIP Request
| |(5060) ------------------> (5060) | |
|25.293 | | BYE | |SIP Request
| | |(5060) ------------------> (5060) |
|25.326 | | 200 OK | |SIP Status
| | |(5060) <------------------ (5060) |
|25.327 | 200 OK | | |SIP Status
| |(5060) <------------------ (5060) | |
Shane
Users mailing list Users@openser.org http://openser.org/cgi-bin/mailman/listinfo/users
I changed the codec to ULAW which is defiantly supported. I'm thinking the reINVITE may be the problem but I'm pretty new at openser configuration and don't see a clear way to detect a reinvite and not auth it. I did capture a ngrep of a failed call. I'll also test xlite or a sipura tonight to see if it something specific to the grandstream.
ngrep -q -t -W byline port 5060 interface: eth0 (192.168.16.0/255.255.255.0) filter: (ip) and ( port 5060 )
U 2007/01/03 16:18:41.764242 192.168.16.91:5060 -> 192.168.16.192:5060 INVITE sip:7005874200@siprt1.siptest.net:5060;user=phone SIP/2.0. t: sip:7005874200@siprt1.siptest.net:5060;user=phone. f: sip:7006311229@192.168.16.91:5060;user=phone;tag=d0b60bd7-1ffd6fd8-5b10359 8. Remote-Party-Id: sip:7006311229@192.168.16.91:5060;user=phone;screen=yes;id-type=subscriber ;party=calling;privacy=off. Proxy-Require: privacy. i: 1068bc1b-1bb-1ffd6fd8@192.168.16.91. CSeq: 12979699 INVITE. v: SIP/2.0/UDP 192.168.16.91:5060. Max-Forwards: 70. m: sip:7006311229@192.168.16.91:5060;user=phone. k: replaces. c: application/sdp. Accept: application/sdp. Accept-Encoding: . Accept-Language: en. User-Agent: MSTSYLVAIPGW. l: 249. . v=0. o=MSTNT 536702936 536702936 IN IP4 192.168.16.91. s=Session SDP. c=IN IP4 192.168.16.91. t=0 0. m=audio 40878 RTP/AVP 18 0 101 . a=silenceSupp:off. a=ecan:b on g168. a=rtpmap:101 telephone-event/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:18 G729/8000.
U 2007/01/03 16:18:41.764646 192.168.16.192:5060 -> 192.168.16.91:5060 SIP/2.0 100 Giving a try. t: sip:7005874200@siprt1.siptest.net:5060;user=phone. f: sip:7006311229@192.168.16.91:5060;user=phone;tag=d0b60bd7-1ffd6fd8-5b10359 8. i: 1068bc1b-1bb-1ffd6fd8@192.168.16.91. CSeq: 12979699 INVITE. v: SIP/2.0/UDP 192.168.16.91:5060. Server: OpenSer (1.2.0-dev12-notls (i386/linux)). Content-Length: 0. Warning: 392 192.168.16.192:5060 "Noisy feedback tells: pid=14845 req_src_ip=192.168.16.91 req_src_port=5060 in_uri=sip:7005874200@siprt1.siptest.net:5060;user=phone out_uri=sip:7005874200@192.168.17.83;user=phone via_cnt==1". .
U 2007/01/03 16:18:41.764673 192.168.16.192:5060 -> 192.168.17.83:5060 INVITE sip:7005874200@192.168.17.83;user=phone SIP/2.0. Record-Route: sip:192.168.16.192;lr=on;ftag=d0b60bd7-1ffd6fd8-5b103598. t: sip:7005874200@siprt1.siptest.net:5060;user=phone. f: sip:7006311229@192.168.16.91:5060;user=phone;tag=d0b60bd7-1ffd6fd8-5b10359 8. Remote-Party-Id: sip:7006311229@192.168.16.91:5060;user=phone;screen=yes;id-type=subscriber ;party=calling;privacy=off. Proxy-Require: privacy. i: 1068bc1b-1bb-1ffd6fd8@192.168.16.91. CSeq: 12979699 INVITE. Via: SIP/2.0/UDP 192.168.16.192;branch=z9hG4bK3f0e.9f290d35.0. v: SIP/2.0/UDP 192.168.16.91:5060. Max-Forwards: 69. m: sip:7006311229@192.168.16.91:5060;user=phone. k: replaces. c: application/sdp. Accept: application/sdp. Accept-Encoding: . Accept-Language: en. User-Agent: MSTSYLVAIPGW. l: 249. . v=0. o=MSTNT 536702936 536702936 IN IP4 192.168.16.91. s=Session SDP. c=IN IP4 192.168.16.91. t=0 0. m=audio 40878 RTP/AVP 18 0 101 . a=silenceSupp:off. a=ecan:b on g168. a=rtpmap:101 telephone-event/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:18 G729/8000.
U 2007/01/03 16:18:41.774319 192.168.17.83:5060 -> 192.168.16.192:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 192.168.16.192;branch=z9hG4bK3f0e.9f290d35.0. Via: SIP/2.0/UDP 192.168.16.91:5060. From: sip:7006311229@192.168.16.91:5060;user=phone;tag=d0b60bd7-1ffd6fd8-5b10359 8. To: sip:7005874200@siprt1.siptest.net:5060;user=phone. Call-ID: 1068bc1b-1bb-1ffd6fd8@192.168.16.91. CSeq: 12979699 INVITE. User-Agent: Grandstream HT496 1.0.3.64 FXS0. Content-Length: 0. .
U 2007/01/03 16:18:41.776363 192.168.17.83:5060 -> 192.168.16.192:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 192.168.16.192;branch=z9hG4bK3f0e.9f290d35.0. Via: SIP/2.0/UDP 192.168.16.91:5060. Record-Route: sip:192.168.16.192;lr=on;ftag=d0b60bd7-1ffd6fd8-5b103598. From: sip:7006311229@192.168.16.91:5060;user=phone;tag=d0b60bd7-1ffd6fd8-5b10359 8. To: sip:7005874200@siprt1.siptest.net:5060;user=phone;tag=6f6a15c679df8aa8. Call-ID: 1068bc1b-1bb-1ffd6fd8@192.168.16.91. CSeq: 12979699 INVITE. User-Agent: Grandstream HT496 1.0.3.64 FXS0. Content-Length: 0. .
U 2007/01/03 16:18:41.776442 192.168.16.192:5060 -> 192.168.16.91:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 192.168.16.91:5060. Record-Route: sip:192.168.16.192;lr=on;ftag=d0b60bd7-1ffd6fd8-5b103598. From: sip:7006311229@192.168.16.91:5060;user=phone;tag=d0b60bd7-1ffd6fd8-5b10359 8. To: sip:7005874200@siprt1.siptest.net:5060;user=phone;tag=6f6a15c679df8aa8. Call-ID: 1068bc1b-1bb-1ffd6fd8@192.168.16.91. CSeq: 12979699 INVITE. User-Agent: Grandstream HT496 1.0.3.64 FXS0. Content-Length: 0. .
U 2007/01/03 16:18:44.426461 192.168.17.83:5060 -> 192.168.16.192:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.16.192;branch=z9hG4bK3f0e.9f290d35.0. Via: SIP/2.0/UDP 192.168.16.91:5060. Record-Route: sip:192.168.16.192;lr=on;ftag=d0b60bd7-1ffd6fd8-5b103598. From: sip:7006311229@192.168.16.91:5060;user=phone;tag=d0b60bd7-1ffd6fd8-5b10359 8. To: sip:7005874200@siprt1.siptest.net:5060;user=phone;tag=6f6a15c679df8aa8. Call-ID: 1068bc1b-1bb-1ffd6fd8@192.168.16.91. CSeq: 12979699 INVITE. User-Agent: Grandstream HT496 1.0.3.64 FXS0. Contact: sip:7005874200@192.168.17.83;user=phone. Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE. Content-Type: application/sdp. Supported: replaces. Content-Length: 220. . v=0. o=7005874200 8000 8000 IN IP4 192.168.17.83. s=SIP Call. c=IN IP4 192.168.17.83. t=0 0. m=audio 10000 RTP/AVP 18 101. a=sendrecv. a=rtpmap:18 G729/8000. a=ptime:20. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-11.
U 2007/01/03 16:18:44.426613 192.168.16.192:5060 -> 192.168.16.91:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.16.91:5060. Record-Route: sip:192.168.16.192;lr=on;ftag=d0b60bd7-1ffd6fd8-5b103598. From: sip:7006311229@192.168.16.91:5060;user=phone;tag=d0b60bd7-1ffd6fd8-5b10359 8. To: sip:7005874200@siprt1.siptest.net:5060;user=phone;tag=6f6a15c679df8aa8. Call-ID: 1068bc1b-1bb-1ffd6fd8@192.168.16.91. CSeq: 12979699 INVITE. User-Agent: Grandstream HT496 1.0.3.64 FXS0. Contact: sip:7005874200@192.168.17.83;user=phone. Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE. Content-Type: application/sdp. Supported: replaces. Content-Length: 220. . v=0. o=7005874200 8000 8000 IN IP4 192.168.17.83. s=SIP Call. c=IN IP4 192.168.17.83. t=0 0. m=audio 10000 RTP/AVP 18 101. a=sendrecv. a=rtpmap:18 G729/8000. a=ptime:20. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-11.
U 2007/01/03 16:18:44.451021 192.168.16.91:5060 -> 192.168.16.192:5060 ACK sip:192.168.16.192;lr=on;ftag=d0b60bd7-1ffd6fd8-5b103598 SIP/2.0. t: sip:7005874200@siprt1.siptest.net:5060;user=phone;tag=6f6a15c679df8aa8. f: sip:7006311229@192.168.16.91:5060;user=phone;tag=d0b60bd7-1ffd6fd8-5b10359 8. i: 1068bc1b-1bb-1ffd6fd8@192.168.16.91. CSeq: 12979699 ACK. v: SIP/2.0/UDP 192.168.16.91:5060. Max-Forwards: 70. Route: sip:7005874200@192.168.17.83;user=phone. User-Agent: MSTSYLVAIPGW. l: 0. .
U 2007/01/03 16:18:44.451221 192.168.16.192:5060 -> 192.168.17.83:5060 ACK sip:7005874200@192.168.17.83;user=phone SIP/2.0. Record-Route: sip:192.168.16.192;lr=on;ftag=d0b60bd7-1ffd6fd8-5b103598. t: sip:7005874200@siprt1.siptest.net:5060;user=phone;tag=6f6a15c679df8aa8. f: sip:7006311229@192.168.16.91:5060;user=phone;tag=d0b60bd7-1ffd6fd8-5b10359 8. i: 1068bc1b-1bb-1ffd6fd8@192.168.16.91. CSeq: 12979699 ACK. Via: SIP/2.0/UDP 192.168.16.192;branch=z9hG4bK3f0e.9f290d35.2. v: SIP/2.0/UDP 192.168.16.91:5060. Max-Forwards: 69. User-Agent: MSTSYLVAIPGW. l: 0. .
U 2007/01/03 16:18:44.451328 192.168.16.91:5060 -> 192.168.16.192:5060 INVITE sip:192.168.16.192;lr=on;ftag=d0b60bd7-1ffd6fd8-5b103598 SIP/2.0. t: sip:7005874200@siprt1.siptest.net:5060;user=phone;tag=6f6a15c679df8aa8. f: sip:7006311229@192.168.16.91:5060;user=phone;tag=d0b60bd7-1ffd6fd8-5b10359 8. Remote-Party-Id: sip:7006311229@192.168.16.91:5060;user=phone;screen=yes;id-type=subscriber ;party=calling;privacy=off. Proxy-Require: privacy. i: 1068bc1b-1bb-1ffd6fd8@192.168.16.91. CSeq: 12979700 INVITE. v: SIP/2.0/UDP 192.168.16.91:5060. Max-Forwards: 70. Route: sip:7005874200@192.168.17.83;user=phone. m: sip:7006311229@192.168.16.91:5060;user=phone. c: application/sdp. Accept: application/sdp. Accept-Encoding: . Accept-Language: en. User-Agent: MSTSYLVAIPGW. l: 236. . v=0. o=MSTNT 536702936 536702937 IN IP4 192.168.16.91. s=Session SDP. c=IN IP4 192.168.16.91. t=0 0. m=audio 40878 RTP/AVP 18 101. a=silenceSupp:off. a=ecan:b on g168. a=ptime:20. a=rtpmap:101 telephone-event/8000. a=rtpmap:18 G729/8000.
U 2007/01/03 16:18:44.453398 192.168.16.192:5060 -> 192.168.16.91:5060 SIP/2.0 407 Proxy Authentication Required. t: sip:7005874200@siprt1.siptest.net:5060;user=phone;tag=6f6a15c679df8aa8. f: sip:7006311229@192.168.16.91:5060;user=phone;tag=d0b60bd7-1ffd6fd8-5b10359 8. i: 1068bc1b-1bb-1ffd6fd8@192.168.16.91. CSeq: 12979700 INVITE. v: SIP/2.0/UDP 192.168.16.91:5060. Proxy-Authenticate: Digest realm="192.168.16.91", nonce="459c1ee0731e3291a5704b9666ffded6acb20bb5". Server: OpenSer (1.2.0-dev12-notls (i386/linux)). Content-Length: 0. Warning: 392 192.168.16.192:5060 "Noisy feedback tells: pid=14844 req_src_ip=192.168.16.91 req_src_port=5060 in_uri=sip:192.168.16.192;lr=on;ftag=d0b60bd7-1ffd6fd8-5b103598 out_uri=sip:7005874200@192.168.17.83;user=phone via_cnt==1". .
U 2007/01/03 16:18:44.471821 192.168.16.91:5060 -> 192.168.16.192:5060 ACK sip:192.168.16.192;lr=on;ftag=d0b60bd7-1ffd6fd8-5b103598 SIP/2.0. t: sip:7005874200@siprt1.siptest.net:5060;user=phone;tag=6f6a15c679df8aa8. f: sip:7006311229@192.168.16.91:5060;user=phone;tag=d0b60bd7-1ffd6fd8-5b10359 8. i: 1068bc1b-1bb-1ffd6fd8@192.168.16.91. CSeq: 12979700 ACK. v: SIP/2.0/UDP 192.168.16.91:5060. Max-Forwards: 70. User-Agent: MSTSYLVAIPGW. l: 0. .
U 2007/01/03 16:18:44.472045 192.168.16.91:5060 -> 192.168.16.192:5060 BYE sip:192.168.16.192;lr=on;ftag=d0b60bd7-1ffd6fd8-5b103598 SIP/2.0. t: sip:7005874200@siprt1.siptest.net:5060;user=phone;tag=6f6a15c679df8aa8. f: sip:7006311229@192.168.16.91:5060;user=phone;tag=d0b60bd7-1ffd6fd8-5b10359 8. i: 1068bc1b-1bb-1ffd6fd8@192.168.16.91. CSeq: 12979701 BYE. v: SIP/2.0/UDP 192.168.16.91:5060. Max-Forwards: 70. Route: sip:7005874200@192.168.17.83;user=phone. Accept: application/sdp. Accept-Encoding: . Accept-Language: en. User-Agent: MSTSYLVAIPGW. l: 0. .
U 2007/01/03 16:18:44.474251 192.168.16.192:5060 -> 192.168.17.83:5060 BYE sip:7005874200@192.168.17.83;user=phone SIP/2.0. Record-Route: sip:192.168.16.192;lr=on;ftag=d0b60bd7-1ffd6fd8-5b103598. t: sip:7005874200@siprt1.siptest.net:5060;user=phone;tag=6f6a15c679df8aa8. f: sip:7006311229@192.168.16.91:5060;user=phone;tag=d0b60bd7-1ffd6fd8-5b10359 8. i: 1068bc1b-1bb-1ffd6fd8@192.168.16.91. CSeq: 12979701 BYE. Via: SIP/2.0/UDP 192.168.16.192;branch=z9hG4bKd29b.7f5f9c52.0. v: SIP/2.0/UDP 192.168.16.91:5060. Max-Forwards: 69. Accept: application/sdp. Accept-Encoding: . Accept-Language: en. User-Agent: MSTSYLVAIPGW. l: 0. .
U 2007/01/03 16:18:44.506433 192.168.17.83:5060 -> 192.168.16.192:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.16.192;branch=z9hG4bKd29b.7f5f9c52.0. Via: SIP/2.0/UDP 192.168.16.91:5060. Record-Route: sip:192.168.16.192;lr=on;ftag=d0b60bd7-1ffd6fd8-5b103598. From: sip:7006311229@192.168.16.91:5060;user=phone;tag=d0b60bd7-1ffd6fd8-5b10359 8. To: sip:7005874200@siprt1.siptest.net:5060;user=phone;tag=6f6a15c679df8aa8. Call-ID: 1068bc1b-1bb-1ffd6fd8@192.168.16.91. CSeq: 12979701 BYE. User-Agent: Grandstream HT496 1.0.3.64 FXS0. Contact: sip:7005874200@192.168.17.83;user=phone. Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE. Supported: replaces. Content-Length: 0. .
U 2007/01/03 16:18:44.506549 192.168.16.192:5060 -> 192.168.16.91:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.16.91:5060. Record-Route: sip:192.168.16.192;lr=on;ftag=d0b60bd7-1ffd6fd8-5b103598. From: sip:7006311229@192.168.16.91:5060;user=phone;tag=d0b60bd7-1ffd6fd8-5b10359 8. To: sip:7005874200@siprt1.siptest.net:5060;user=phone;tag=6f6a15c679df8aa8. Call-ID: 1068bc1b-1bb-1ffd6fd8@192.168.16.91. CSeq: 12979701 BYE. User-Agent: Grandstream HT496 1.0.3.64 FXS0. Contact: sip:7005874200@192.168.17.83;user=phone. Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE. Supported: replaces. Content-Length: 0. .
U 2007/01/03 16:18:47.652882 192.168.17.150:5060 -> 192.168.16.192:5060 ..................
-----Original Message----- From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] Sent: Wednesday, January 03, 2007 7:51 AM To: Shane Burrell Cc: users@openser.org Subject: Re: [Users] Issues with calls using openser.
The log shows that you challenge reINVITEs. MAybe this breaks the grandstream. Please try without challenging the reINVITE. If this helps, then it is a probably a grandstream bug.
But of course there is the question why the grandstream sends a reINVITE at all? This is often a codec problem.
Please also try to use other client (e.g. xlite) and post complete ngrep dumps: "ngrep -q -t -W byline port 5060"
regards klaus
Shane Burrell wrote:
The SIP UA was a grandstream ATA running the latest stable firmware.
Prior
to upgrading to 1.1 and moving to mediaproxy it worked well with the exception of good nat support which is why I would really like mediaproxy
to
work. Is there anything I should look for in the sip dialog to determine
if
the client, sip proxy, or the gateway is the culprit on disconnecting the call?
Thanks,
Shane
-----Original Message----- From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] Sent: Wednesday, January 03, 2007 4:52 AM To: Shane Burrell Cc: users@openser.org Subject: Re: [Users] Issues with calls using openser.
Maybe a bug in the caller's SIP client?
regards klaus
Shane Burrell wrote:
I recently installed the latest version of openser and this time used mediaproxy rather than rtpproxy. Everything seems to work but if a sip device is called the phone rings and is instantally disconnected and the
far
end is left off-hook. This worked before but I did modify my script to
work
with mediaproxy. Below is the wireshark decode of the sip messagining.
Any
help or suggestions on where to look would be great. Calls from the sip device works flawlessly. I am using a MaxTNT as the gateway.
|Time | 152.93.36.91 | siprt1.me.net| 152.93.37.83 |
|22.031 | INVITE SDP ( telephone-event) | |SIP From: sip:8385021101@192.168.16.91:5060 To:sip: 8385024200@ siprt1.me.net:5060
| |(5060) ------------------> (5060) | |
|22.031 | 100 Giving a try |
|SIP
Status
| |(5060) <------------------ (5060) | |
|22.031 | | INVITE SDP ( telephone-event) |SIP Request
| | |(5060) ------------------> (5060) |
|22.040 | | 100 Trying|
|SIP
Status
| | |(5060) <------------------ (5060) |
|22.042 | | 180 Ringing
|SIP
Status
| | |(5060) <------------------ (5060) |
|22.042 | 180 Ringing |
|SIP
Status
| |(5060) <------------------ (5060) | |
|25.244 | | 200 OK SDP ( telephone-event) |SIP Status
| | |(5060) <------------------ (5060) |
|25.245 | 200 OK SDP ( telephone-event) | |SIP Status
| |(5060) <------------------ (5060) | |
|25.269 | ACK | |
|SIP
Request
| |(5060) ------------------> (5060) | |
|25.269 | | ACK |
|SIP
Request
| | |(5060) ------------------> (5060) |
|25.269 | INVITE SDP ( telephone-event) | |SIP From: sip: 8385021101@152.93.36.91:5060 To:sip: 8385024200@ siprt1.me.net:5060
| |(5060) ------------------> (5060) | |
|25.270 | 407 Proxy Authentication Required | |SIP Status
| |(5060) <------------------ (5060) | |
|25.291 | ACK | |
|SIP
Request
| |(5060) ------------------> (5060) | |
|25.291 | BYE | |
|SIP
Request
| |(5060) ------------------> (5060) | |
|25.293 | | BYE |
|SIP
Request
| | |(5060) ------------------> (5060) |
|25.326 | | 200 OK |
|SIP
Status
| | |(5060) <------------------ (5060) |
|25.327 | 200 OK | |
|SIP
Status
| |(5060) <------------------ (5060) | |
Shane
Users mailing list Users@openser.org http://openser.org/cgi-bin/mailman/listinfo/users
Hi!
I think there are several problems:
1. your config is broken. It looks like the caller is a gateway. Usually gateways are not authenticated by the proxy with digest authentication but use IP based authentication. Thus, as you challenge the reINVITE, the gateway has no credentials to answer the challenge and hangs up.
2. The gateway offers µ-LAW and G.729. The Grandstream answers with G.729. This looks like a normal call setup. Nevertheless the gateway sends a reINVITE with explicitely offering only G.729. Why? Maybe the gateway does not support receiving µ-LAW and sending G.729 - thus it sends a reINVITE to force the received codec to G.729.
Conclusion - fix your config.
Use IP based authentication and TCP (e.g. is_fromgw from LCR module) to authenticate your gateway to the proxy. Use digest authentication to authenticate users.
reINVITEs (which are handled in the loose_route() block) can be also authenticated using the above methods - but not authenticating reINVITEs is usually no security issue as the totag acts somehow as a "cookie" to the UAS.
regards klaus
On Wed, January 3, 2007 22:25, Shane Burrell said:
I changed the codec to ULAW which is defiantly supported. I'm thinking the reINVITE may be the problem but I'm pretty new at openser configuration and don't see a clear way to detect a reinvite and not auth it. I did capture a ngrep of a failed call. I'll also test xlite or a sipura tonight to see if it something specific to the grandstream.
ngrep -q -t -W byline port 5060 interface: eth0 (192.168.16.0/255.255.255.0) filter: (ip) and ( port 5060 )
U 2007/01/03 16:18:41.764242 192.168.16.91:5060 -> 192.168.16.192:5060 INVITE sip:7005874200@siprt1.siptest.net:5060;user=phone SIP/2.0. t: sip:7005874200@siprt1.siptest.net:5060;user=phone. f: sip:7006311229@192.168.16.91:5060;user=phone;tag=d0b60bd7-1ffd6fd8-5b10359 8. Remote-Party-Id: sip:7006311229@192.168.16.91:5060;user=phone;screen=yes;id-type=subscriber ;party=calling;privacy=off. Proxy-Require: privacy. i: 1068bc1b-1bb-1ffd6fd8@192.168.16.91. CSeq: 12979699 INVITE. v: SIP/2.0/UDP 192.168.16.91:5060. Max-Forwards: 70. m: sip:7006311229@192.168.16.91:5060;user=phone. k: replaces. c: application/sdp. Accept: application/sdp. Accept-Encoding: . Accept-Language: en. User-Agent: MSTSYLVAIPGW. l: 249. . v=0. o=MSTNT 536702936 536702936 IN IP4 192.168.16.91. s=Session SDP. c=IN IP4 192.168.16.91. t=0 0. m=audio 40878 RTP/AVP 18 0 101 . a=silenceSupp:off. a=ecan:b on g168. a=rtpmap:101 telephone-event/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:18 G729/8000.
U 2007/01/03 16:18:41.764646 192.168.16.192:5060 -> 192.168.16.91:5060 SIP/2.0 100 Giving a try. t: sip:7005874200@siprt1.siptest.net:5060;user=phone. f: sip:7006311229@192.168.16.91:5060;user=phone;tag=d0b60bd7-1ffd6fd8-5b10359 8. i: 1068bc1b-1bb-1ffd6fd8@192.168.16.91. CSeq: 12979699 INVITE. v: SIP/2.0/UDP 192.168.16.91:5060. Server: OpenSer (1.2.0-dev12-notls (i386/linux)). Content-Length: 0. Warning: 392 192.168.16.192:5060 "Noisy feedback tells: pid=14845 req_src_ip=192.168.16.91 req_src_port=5060 in_uri=sip:7005874200@siprt1.siptest.net:5060;user=phone out_uri=sip:7005874200@192.168.17.83;user=phone via_cnt==1". .
U 2007/01/03 16:18:41.764673 192.168.16.192:5060 -> 192.168.17.83:5060 INVITE sip:7005874200@192.168.17.83;user=phone SIP/2.0. Record-Route: sip:192.168.16.192;lr=on;ftag=d0b60bd7-1ffd6fd8-5b103598. t: sip:7005874200@siprt1.siptest.net:5060;user=phone. f: sip:7006311229@192.168.16.91:5060;user=phone;tag=d0b60bd7-1ffd6fd8-5b10359 8. Remote-Party-Id: sip:7006311229@192.168.16.91:5060;user=phone;screen=yes;id-type=subscriber ;party=calling;privacy=off. Proxy-Require: privacy. i: 1068bc1b-1bb-1ffd6fd8@192.168.16.91. CSeq: 12979699 INVITE. Via: SIP/2.0/UDP 192.168.16.192;branch=z9hG4bK3f0e.9f290d35.0. v: SIP/2.0/UDP 192.168.16.91:5060. Max-Forwards: 69. m: sip:7006311229@192.168.16.91:5060;user=phone. k: replaces. c: application/sdp. Accept: application/sdp. Accept-Encoding: . Accept-Language: en. User-Agent: MSTSYLVAIPGW. l: 249. . v=0. o=MSTNT 536702936 536702936 IN IP4 192.168.16.91. s=Session SDP. c=IN IP4 192.168.16.91. t=0 0. m=audio 40878 RTP/AVP 18 0 101 . a=silenceSupp:off. a=ecan:b on g168. a=rtpmap:101 telephone-event/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:18 G729/8000.
U 2007/01/03 16:18:41.774319 192.168.17.83:5060 -> 192.168.16.192:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 192.168.16.192;branch=z9hG4bK3f0e.9f290d35.0. Via: SIP/2.0/UDP 192.168.16.91:5060. From: sip:7006311229@192.168.16.91:5060;user=phone;tag=d0b60bd7-1ffd6fd8-5b10359 8. To: sip:7005874200@siprt1.siptest.net:5060;user=phone. Call-ID: 1068bc1b-1bb-1ffd6fd8@192.168.16.91. CSeq: 12979699 INVITE. User-Agent: Grandstream HT496 1.0.3.64 FXS0. Content-Length: 0. .
U 2007/01/03 16:18:41.776363 192.168.17.83:5060 -> 192.168.16.192:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 192.168.16.192;branch=z9hG4bK3f0e.9f290d35.0. Via: SIP/2.0/UDP 192.168.16.91:5060. Record-Route: sip:192.168.16.192;lr=on;ftag=d0b60bd7-1ffd6fd8-5b103598. From: sip:7006311229@192.168.16.91:5060;user=phone;tag=d0b60bd7-1ffd6fd8-5b10359 8. To: sip:7005874200@siprt1.siptest.net:5060;user=phone;tag=6f6a15c679df8aa8. Call-ID: 1068bc1b-1bb-1ffd6fd8@192.168.16.91. CSeq: 12979699 INVITE. User-Agent: Grandstream HT496 1.0.3.64 FXS0. Content-Length: 0. .
U 2007/01/03 16:18:41.776442 192.168.16.192:5060 -> 192.168.16.91:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 192.168.16.91:5060. Record-Route: sip:192.168.16.192;lr=on;ftag=d0b60bd7-1ffd6fd8-5b103598. From: sip:7006311229@192.168.16.91:5060;user=phone;tag=d0b60bd7-1ffd6fd8-5b10359 8. To: sip:7005874200@siprt1.siptest.net:5060;user=phone;tag=6f6a15c679df8aa8. Call-ID: 1068bc1b-1bb-1ffd6fd8@192.168.16.91. CSeq: 12979699 INVITE. User-Agent: Grandstream HT496 1.0.3.64 FXS0. Content-Length: 0. .
U 2007/01/03 16:18:44.426461 192.168.17.83:5060 -> 192.168.16.192:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.16.192;branch=z9hG4bK3f0e.9f290d35.0. Via: SIP/2.0/UDP 192.168.16.91:5060. Record-Route: sip:192.168.16.192;lr=on;ftag=d0b60bd7-1ffd6fd8-5b103598. From: sip:7006311229@192.168.16.91:5060;user=phone;tag=d0b60bd7-1ffd6fd8-5b10359 8. To: sip:7005874200@siprt1.siptest.net:5060;user=phone;tag=6f6a15c679df8aa8. Call-ID: 1068bc1b-1bb-1ffd6fd8@192.168.16.91. CSeq: 12979699 INVITE. User-Agent: Grandstream HT496 1.0.3.64 FXS0. Contact: sip:7005874200@192.168.17.83;user=phone. Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE. Content-Type: application/sdp. Supported: replaces. Content-Length: 220. . v=0. o=7005874200 8000 8000 IN IP4 192.168.17.83. s=SIP Call. c=IN IP4 192.168.17.83. t=0 0. m=audio 10000 RTP/AVP 18 101. a=sendrecv. a=rtpmap:18 G729/8000. a=ptime:20. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-11.
U 2007/01/03 16:18:44.426613 192.168.16.192:5060 -> 192.168.16.91:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.16.91:5060. Record-Route: sip:192.168.16.192;lr=on;ftag=d0b60bd7-1ffd6fd8-5b103598. From: sip:7006311229@192.168.16.91:5060;user=phone;tag=d0b60bd7-1ffd6fd8-5b10359 8. To: sip:7005874200@siprt1.siptest.net:5060;user=phone;tag=6f6a15c679df8aa8. Call-ID: 1068bc1b-1bb-1ffd6fd8@192.168.16.91. CSeq: 12979699 INVITE. User-Agent: Grandstream HT496 1.0.3.64 FXS0. Contact: sip:7005874200@192.168.17.83;user=phone. Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE. Content-Type: application/sdp. Supported: replaces. Content-Length: 220. . v=0. o=7005874200 8000 8000 IN IP4 192.168.17.83. s=SIP Call. c=IN IP4 192.168.17.83. t=0 0. m=audio 10000 RTP/AVP 18 101. a=sendrecv. a=rtpmap:18 G729/8000. a=ptime:20. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-11.
U 2007/01/03 16:18:44.451021 192.168.16.91:5060 -> 192.168.16.192:5060 ACK sip:192.168.16.192;lr=on;ftag=d0b60bd7-1ffd6fd8-5b103598 SIP/2.0. t: sip:7005874200@siprt1.siptest.net:5060;user=phone;tag=6f6a15c679df8aa8. f: sip:7006311229@192.168.16.91:5060;user=phone;tag=d0b60bd7-1ffd6fd8-5b10359 8. i: 1068bc1b-1bb-1ffd6fd8@192.168.16.91. CSeq: 12979699 ACK. v: SIP/2.0/UDP 192.168.16.91:5060. Max-Forwards: 70. Route: sip:7005874200@192.168.17.83;user=phone. User-Agent: MSTSYLVAIPGW. l: 0. .
U 2007/01/03 16:18:44.451221 192.168.16.192:5060 -> 192.168.17.83:5060 ACK sip:7005874200@192.168.17.83;user=phone SIP/2.0. Record-Route: sip:192.168.16.192;lr=on;ftag=d0b60bd7-1ffd6fd8-5b103598. t: sip:7005874200@siprt1.siptest.net:5060;user=phone;tag=6f6a15c679df8aa8. f: sip:7006311229@192.168.16.91:5060;user=phone;tag=d0b60bd7-1ffd6fd8-5b10359 8. i: 1068bc1b-1bb-1ffd6fd8@192.168.16.91. CSeq: 12979699 ACK. Via: SIP/2.0/UDP 192.168.16.192;branch=z9hG4bK3f0e.9f290d35.2. v: SIP/2.0/UDP 192.168.16.91:5060. Max-Forwards: 69. User-Agent: MSTSYLVAIPGW. l: 0. .
U 2007/01/03 16:18:44.451328 192.168.16.91:5060 -> 192.168.16.192:5060 INVITE sip:192.168.16.192;lr=on;ftag=d0b60bd7-1ffd6fd8-5b103598 SIP/2.0. t: sip:7005874200@siprt1.siptest.net:5060;user=phone;tag=6f6a15c679df8aa8. f: sip:7006311229@192.168.16.91:5060;user=phone;tag=d0b60bd7-1ffd6fd8-5b10359 8. Remote-Party-Id: sip:7006311229@192.168.16.91:5060;user=phone;screen=yes;id-type=subscriber ;party=calling;privacy=off. Proxy-Require: privacy. i: 1068bc1b-1bb-1ffd6fd8@192.168.16.91. CSeq: 12979700 INVITE. v: SIP/2.0/UDP 192.168.16.91:5060. Max-Forwards: 70. Route: sip:7005874200@192.168.17.83;user=phone. m: sip:7006311229@192.168.16.91:5060;user=phone. c: application/sdp. Accept: application/sdp. Accept-Encoding: . Accept-Language: en. User-Agent: MSTSYLVAIPGW. l: 236. . v=0. o=MSTNT 536702936 536702937 IN IP4 192.168.16.91. s=Session SDP. c=IN IP4 192.168.16.91. t=0 0. m=audio 40878 RTP/AVP 18 101. a=silenceSupp:off. a=ecan:b on g168. a=ptime:20. a=rtpmap:101 telephone-event/8000. a=rtpmap:18 G729/8000.
U 2007/01/03 16:18:44.453398 192.168.16.192:5060 -> 192.168.16.91:5060 SIP/2.0 407 Proxy Authentication Required. t: sip:7005874200@siprt1.siptest.net:5060;user=phone;tag=6f6a15c679df8aa8. f: sip:7006311229@192.168.16.91:5060;user=phone;tag=d0b60bd7-1ffd6fd8-5b10359 8. i: 1068bc1b-1bb-1ffd6fd8@192.168.16.91. CSeq: 12979700 INVITE. v: SIP/2.0/UDP 192.168.16.91:5060. Proxy-Authenticate: Digest realm="192.168.16.91", nonce="459c1ee0731e3291a5704b9666ffded6acb20bb5". Server: OpenSer (1.2.0-dev12-notls (i386/linux)). Content-Length: 0. Warning: 392 192.168.16.192:5060 "Noisy feedback tells: pid=14844 req_src_ip=192.168.16.91 req_src_port=5060 in_uri=sip:192.168.16.192;lr=on;ftag=d0b60bd7-1ffd6fd8-5b103598 out_uri=sip:7005874200@192.168.17.83;user=phone via_cnt==1". .
U 2007/01/03 16:18:44.471821 192.168.16.91:5060 -> 192.168.16.192:5060 ACK sip:192.168.16.192;lr=on;ftag=d0b60bd7-1ffd6fd8-5b103598 SIP/2.0. t: sip:7005874200@siprt1.siptest.net:5060;user=phone;tag=6f6a15c679df8aa8. f: sip:7006311229@192.168.16.91:5060;user=phone;tag=d0b60bd7-1ffd6fd8-5b10359 8. i: 1068bc1b-1bb-1ffd6fd8@192.168.16.91. CSeq: 12979700 ACK. v: SIP/2.0/UDP 192.168.16.91:5060. Max-Forwards: 70. User-Agent: MSTSYLVAIPGW. l: 0. .
U 2007/01/03 16:18:44.472045 192.168.16.91:5060 -> 192.168.16.192:5060 BYE sip:192.168.16.192;lr=on;ftag=d0b60bd7-1ffd6fd8-5b103598 SIP/2.0. t: sip:7005874200@siprt1.siptest.net:5060;user=phone;tag=6f6a15c679df8aa8. f: sip:7006311229@192.168.16.91:5060;user=phone;tag=d0b60bd7-1ffd6fd8-5b10359 8. i: 1068bc1b-1bb-1ffd6fd8@192.168.16.91. CSeq: 12979701 BYE. v: SIP/2.0/UDP 192.168.16.91:5060. Max-Forwards: 70. Route: sip:7005874200@192.168.17.83;user=phone. Accept: application/sdp. Accept-Encoding: . Accept-Language: en. User-Agent: MSTSYLVAIPGW. l: 0. .
U 2007/01/03 16:18:44.474251 192.168.16.192:5060 -> 192.168.17.83:5060 BYE sip:7005874200@192.168.17.83;user=phone SIP/2.0. Record-Route: sip:192.168.16.192;lr=on;ftag=d0b60bd7-1ffd6fd8-5b103598. t: sip:7005874200@siprt1.siptest.net:5060;user=phone;tag=6f6a15c679df8aa8. f: sip:7006311229@192.168.16.91:5060;user=phone;tag=d0b60bd7-1ffd6fd8-5b10359 8. i: 1068bc1b-1bb-1ffd6fd8@192.168.16.91. CSeq: 12979701 BYE. Via: SIP/2.0/UDP 192.168.16.192;branch=z9hG4bKd29b.7f5f9c52.0. v: SIP/2.0/UDP 192.168.16.91:5060. Max-Forwards: 69. Accept: application/sdp. Accept-Encoding: . Accept-Language: en. User-Agent: MSTSYLVAIPGW. l: 0. .
U 2007/01/03 16:18:44.506433 192.168.17.83:5060 -> 192.168.16.192:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.16.192;branch=z9hG4bKd29b.7f5f9c52.0. Via: SIP/2.0/UDP 192.168.16.91:5060. Record-Route: sip:192.168.16.192;lr=on;ftag=d0b60bd7-1ffd6fd8-5b103598. From: sip:7006311229@192.168.16.91:5060;user=phone;tag=d0b60bd7-1ffd6fd8-5b10359 8. To: sip:7005874200@siprt1.siptest.net:5060;user=phone;tag=6f6a15c679df8aa8. Call-ID: 1068bc1b-1bb-1ffd6fd8@192.168.16.91. CSeq: 12979701 BYE. User-Agent: Grandstream HT496 1.0.3.64 FXS0. Contact: sip:7005874200@192.168.17.83;user=phone. Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE. Supported: replaces. Content-Length: 0. .
U 2007/01/03 16:18:44.506549 192.168.16.192:5060 -> 192.168.16.91:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.16.91:5060. Record-Route: sip:192.168.16.192;lr=on;ftag=d0b60bd7-1ffd6fd8-5b103598. From: sip:7006311229@192.168.16.91:5060;user=phone;tag=d0b60bd7-1ffd6fd8-5b10359 8. To: sip:7005874200@siprt1.siptest.net:5060;user=phone;tag=6f6a15c679df8aa8. Call-ID: 1068bc1b-1bb-1ffd6fd8@192.168.16.91. CSeq: 12979701 BYE. User-Agent: Grandstream HT496 1.0.3.64 FXS0. Contact: sip:7005874200@192.168.17.83;user=phone. Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE. Supported: replaces. Content-Length: 0. .
U 2007/01/03 16:18:47.652882 192.168.17.150:5060 -> 192.168.16.192:5060 ..................
-----Original Message----- From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] Sent: Wednesday, January 03, 2007 7:51 AM To: Shane Burrell Cc: users@openser.org Subject: Re: [Users] Issues with calls using openser.
The log shows that you challenge reINVITEs. MAybe this breaks the grandstream. Please try without challenging the reINVITE. If this helps, then it is a probably a grandstream bug.
But of course there is the question why the grandstream sends a reINVITE at all? This is often a codec problem.
Please also try to use other client (e.g. xlite) and post complete ngrep dumps: "ngrep -q -t -W byline port 5060"
regards klaus
Shane Burrell wrote:
The SIP UA was a grandstream ATA running the latest stable firmware.
Prior
to upgrading to 1.1 and moving to mediaproxy it worked well with the exception of good nat support which is why I would really like mediaproxy
to
work. Is there anything I should look for in the sip dialog to determine
if
the client, sip proxy, or the gateway is the culprit on disconnecting the call?
Thanks,
Shane
-----Original Message----- From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] Sent: Wednesday, January 03, 2007 4:52 AM To: Shane Burrell Cc: users@openser.org Subject: Re: [Users] Issues with calls using openser.
Maybe a bug in the caller's SIP client?
regards klaus
Shane Burrell wrote:
I recently installed the latest version of openser and this time used mediaproxy rather than rtpproxy. Everything seems to work but if a sip device is called the phone rings and is instantally disconnected and the
far
end is left off-hook. This worked before but I did modify my script to
work
with mediaproxy. Below is the wireshark decode of the sip messagining.
Any
help or suggestions on where to look would be great. Calls from the sip device works flawlessly. I am using a MaxTNT as the gateway.
|Time | 152.93.36.91 | siprt1.me.net| 152.93.37.83 |
|22.031 | INVITE SDP ( telephone-event) | |SIP From: sip:8385021101@192.168.16.91:5060 To:sip: 8385024200@ siprt1.me.net:5060
| |(5060) ------------------> (5060) | |
|22.031 | 100 Giving a try |
|SIP
Status
| |(5060) <------------------ (5060) | |
|22.031 | | INVITE SDP ( telephone-event) |SIP Request
| | |(5060) ------------------> (5060) |
|22.040 | | 100 Trying|
|SIP
Status
| | |(5060) <------------------ (5060) |
|22.042 | | 180 Ringing
|SIP
Status
| | |(5060) <------------------ (5060) |
|22.042 | 180 Ringing |
|SIP
Status
| |(5060) <------------------ (5060) | |
|25.244 | | 200 OK SDP ( telephone-event) |SIP Status
| | |(5060) <------------------ (5060) |
|25.245 | 200 OK SDP ( telephone-event) | |SIP Status
| |(5060) <------------------ (5060) | |
|25.269 | ACK | |
|SIP
Request
| |(5060) ------------------> (5060) | |
|25.269 | | ACK |
|SIP
Request
| | |(5060) ------------------> (5060) |
|25.269 | INVITE SDP ( telephone-event) | |SIP From: sip: 8385021101@152.93.36.91:5060 To:sip: 8385024200@ siprt1.me.net:5060
| |(5060) ------------------> (5060) | |
|25.270 | 407 Proxy Authentication Required | |SIP Status
| |(5060) <------------------ (5060) | |
|25.291 | ACK | |
|SIP
Request
| |(5060) ------------------> (5060) | |
|25.291 | BYE | |
|SIP
Request
| |(5060) ------------------> (5060) | |
|25.293 | | BYE |
|SIP
Request
| | |(5060) ------------------> (5060) |
|25.326 | | 200 OK |
|SIP
Status
| | |(5060) <------------------ (5060) |
|25.327 | 200 OK | |
|SIP
Status
| |(5060) <------------------ (5060) | |
Shane
Users mailing list Users@openser.org http://openser.org/cgi-bin/mailman/listinfo/users
-- Klaus Darilion nic.at
Klaus, Thanks so much for that excellent analysis of my problem. You hit it perfectly. I fixed the problem by added code to always allow the gateway IP access and it works flawlessly.
Thanks again,
Shane
-----Original Message----- From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] Sent: Wednesday, January 03, 2007 5:51 PM To: Shane Burrell Cc: users@openser.org Subject: RE: [Users] Issues with calls using openser.
Hi!
I think there are several problems:
1. your config is broken. It looks like the caller is a gateway. Usually gateways are not authenticated by the proxy with digest authentication but use IP based authentication. Thus, as you challenge the reINVITE, the gateway has no credentials to answer the challenge and hangs up.
2. The gateway offers µ-LAW and G.729. The Grandstream answers with G.729. This looks like a normal call setup. Nevertheless the gateway sends a reINVITE with explicitely offering only G.729. Why? Maybe the gateway does not support receiving µ-LAW and sending G.729 - thus it sends a reINVITE to force the received codec to G.729.
Conclusion - fix your config.
Use IP based authentication and TCP (e.g. is_fromgw from LCR module) to authenticate your gateway to the proxy. Use digest authentication to authenticate users.
reINVITEs (which are handled in the loose_route() block) can be also authenticated using the above methods - but not authenticating reINVITEs is usually no security issue as the totag acts somehow as a "cookie" to the UAS.
regards klaus
On Wed, January 3, 2007 22:25, Shane Burrell said:
I changed the codec to ULAW which is defiantly supported. I'm thinking the reINVITE may be the problem but I'm pretty new at openser configuration and don't see a clear way to detect a reinvite and not auth it. I did capture a ngrep of a failed call. I'll also test xlite or a sipura tonight to see if it something specific to the grandstream.
ngrep -q -t -W byline port 5060 interface: eth0 (192.168.16.0/255.255.255.0) filter: (ip) and ( port 5060 )
U 2007/01/03 16:18:41.764242 192.168.16.91:5060 -> 192.168.16.192:5060 INVITE sip:7005874200@siprt1.siptest.net:5060;user=phone SIP/2.0. t: sip:7005874200@siprt1.siptest.net:5060;user=phone. f:
sip:7006311229@192.168.16.91:5060;user=phone;tag=d0b60bd7-1ffd6fd8-5b10359
Remote-Party-Id:
sip:7006311229@192.168.16.91:5060;user=phone;screen=yes;id-type=subscriber
;party=calling;privacy=off. Proxy-Require: privacy. i: 1068bc1b-1bb-1ffd6fd8@192.168.16.91. CSeq: 12979699 INVITE. v: SIP/2.0/UDP 192.168.16.91:5060. Max-Forwards: 70. m: sip:7006311229@192.168.16.91:5060;user=phone. k: replaces. c: application/sdp. Accept: application/sdp. Accept-Encoding: . Accept-Language: en. User-Agent: MSTSYLVAIPGW. l: 249. . v=0. o=MSTNT 536702936 536702936 IN IP4 192.168.16.91. s=Session SDP. c=IN IP4 192.168.16.91. t=0 0. m=audio 40878 RTP/AVP 18 0 101 . a=silenceSupp:off. a=ecan:b on g168. a=rtpmap:101 telephone-event/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:18 G729/8000.
U 2007/01/03 16:18:41.764646 192.168.16.192:5060 -> 192.168.16.91:5060 SIP/2.0 100 Giving a try. t: sip:7005874200@siprt1.siptest.net:5060;user=phone. f:
sip:7006311229@192.168.16.91:5060;user=phone;tag=d0b60bd7-1ffd6fd8-5b10359
i: 1068bc1b-1bb-1ffd6fd8@192.168.16.91. CSeq: 12979699 INVITE. v: SIP/2.0/UDP 192.168.16.91:5060. Server: OpenSer (1.2.0-dev12-notls (i386/linux)). Content-Length: 0. Warning: 392 192.168.16.192:5060 "Noisy feedback tells: pid=14845 req_src_ip=192.168.16.91 req_src_port=5060 in_uri=sip:7005874200@siprt1.siptest.net:5060;user=phone out_uri=sip:7005874200@192.168.17.83;user=phone via_cnt==1". .
U 2007/01/03 16:18:41.764673 192.168.16.192:5060 -> 192.168.17.83:5060 INVITE sip:7005874200@192.168.17.83;user=phone SIP/2.0. Record-Route: sip:192.168.16.192;lr=on;ftag=d0b60bd7-1ffd6fd8-5b103598. t: sip:7005874200@siprt1.siptest.net:5060;user=phone. f:
sip:7006311229@192.168.16.91:5060;user=phone;tag=d0b60bd7-1ffd6fd8-5b10359
Remote-Party-Id:
sip:7006311229@192.168.16.91:5060;user=phone;screen=yes;id-type=subscriber
;party=calling;privacy=off. Proxy-Require: privacy. i: 1068bc1b-1bb-1ffd6fd8@192.168.16.91. CSeq: 12979699 INVITE. Via: SIP/2.0/UDP 192.168.16.192;branch=z9hG4bK3f0e.9f290d35.0. v: SIP/2.0/UDP 192.168.16.91:5060. Max-Forwards: 69. m: sip:7006311229@192.168.16.91:5060;user=phone. k: replaces. c: application/sdp. Accept: application/sdp. Accept-Encoding: . Accept-Language: en. User-Agent: MSTSYLVAIPGW. l: 249. . v=0. o=MSTNT 536702936 536702936 IN IP4 192.168.16.91. s=Session SDP. c=IN IP4 192.168.16.91. t=0 0. m=audio 40878 RTP/AVP 18 0 101 . a=silenceSupp:off. a=ecan:b on g168. a=rtpmap:101 telephone-event/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:18 G729/8000.
U 2007/01/03 16:18:41.774319 192.168.17.83:5060 -> 192.168.16.192:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 192.168.16.192;branch=z9hG4bK3f0e.9f290d35.0. Via: SIP/2.0/UDP 192.168.16.91:5060. From:
sip:7006311229@192.168.16.91:5060;user=phone;tag=d0b60bd7-1ffd6fd8-5b10359
To: sip:7005874200@siprt1.siptest.net:5060;user=phone. Call-ID: 1068bc1b-1bb-1ffd6fd8@192.168.16.91. CSeq: 12979699 INVITE. User-Agent: Grandstream HT496 1.0.3.64 FXS0. Content-Length: 0. .
U 2007/01/03 16:18:41.776363 192.168.17.83:5060 -> 192.168.16.192:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 192.168.16.192;branch=z9hG4bK3f0e.9f290d35.0. Via: SIP/2.0/UDP 192.168.16.91:5060. Record-Route: sip:192.168.16.192;lr=on;ftag=d0b60bd7-1ffd6fd8-5b103598. From:
sip:7006311229@192.168.16.91:5060;user=phone;tag=d0b60bd7-1ffd6fd8-5b10359
To: sip:7005874200@siprt1.siptest.net:5060;user=phone;tag=6f6a15c679df8aa8. Call-ID: 1068bc1b-1bb-1ffd6fd8@192.168.16.91. CSeq: 12979699 INVITE. User-Agent: Grandstream HT496 1.0.3.64 FXS0. Content-Length: 0. .
U 2007/01/03 16:18:41.776442 192.168.16.192:5060 -> 192.168.16.91:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 192.168.16.91:5060. Record-Route: sip:192.168.16.192;lr=on;ftag=d0b60bd7-1ffd6fd8-5b103598. From:
sip:7006311229@192.168.16.91:5060;user=phone;tag=d0b60bd7-1ffd6fd8-5b10359
To: sip:7005874200@siprt1.siptest.net:5060;user=phone;tag=6f6a15c679df8aa8. Call-ID: 1068bc1b-1bb-1ffd6fd8@192.168.16.91. CSeq: 12979699 INVITE. User-Agent: Grandstream HT496 1.0.3.64 FXS0. Content-Length: 0. .
U 2007/01/03 16:18:44.426461 192.168.17.83:5060 -> 192.168.16.192:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.16.192;branch=z9hG4bK3f0e.9f290d35.0. Via: SIP/2.0/UDP 192.168.16.91:5060. Record-Route: sip:192.168.16.192;lr=on;ftag=d0b60bd7-1ffd6fd8-5b103598. From:
sip:7006311229@192.168.16.91:5060;user=phone;tag=d0b60bd7-1ffd6fd8-5b10359
To: sip:7005874200@siprt1.siptest.net:5060;user=phone;tag=6f6a15c679df8aa8. Call-ID: 1068bc1b-1bb-1ffd6fd8@192.168.16.91. CSeq: 12979699 INVITE. User-Agent: Grandstream HT496 1.0.3.64 FXS0. Contact: sip:7005874200@192.168.17.83;user=phone. Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE. Content-Type: application/sdp. Supported: replaces. Content-Length: 220. . v=0. o=7005874200 8000 8000 IN IP4 192.168.17.83. s=SIP Call. c=IN IP4 192.168.17.83. t=0 0. m=audio 10000 RTP/AVP 18 101. a=sendrecv. a=rtpmap:18 G729/8000. a=ptime:20. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-11.
U 2007/01/03 16:18:44.426613 192.168.16.192:5060 -> 192.168.16.91:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.16.91:5060. Record-Route: sip:192.168.16.192;lr=on;ftag=d0b60bd7-1ffd6fd8-5b103598. From:
sip:7006311229@192.168.16.91:5060;user=phone;tag=d0b60bd7-1ffd6fd8-5b10359
To: sip:7005874200@siprt1.siptest.net:5060;user=phone;tag=6f6a15c679df8aa8. Call-ID: 1068bc1b-1bb-1ffd6fd8@192.168.16.91. CSeq: 12979699 INVITE. User-Agent: Grandstream HT496 1.0.3.64 FXS0. Contact: sip:7005874200@192.168.17.83;user=phone. Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE. Content-Type: application/sdp. Supported: replaces. Content-Length: 220. . v=0. o=7005874200 8000 8000 IN IP4 192.168.17.83. s=SIP Call. c=IN IP4 192.168.17.83. t=0 0. m=audio 10000 RTP/AVP 18 101. a=sendrecv. a=rtpmap:18 G729/8000. a=ptime:20. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-11.
U 2007/01/03 16:18:44.451021 192.168.16.91:5060 -> 192.168.16.192:5060 ACK sip:192.168.16.192;lr=on;ftag=d0b60bd7-1ffd6fd8-5b103598 SIP/2.0. t: sip:7005874200@siprt1.siptest.net:5060;user=phone;tag=6f6a15c679df8aa8. f:
sip:7006311229@192.168.16.91:5060;user=phone;tag=d0b60bd7-1ffd6fd8-5b10359
i: 1068bc1b-1bb-1ffd6fd8@192.168.16.91. CSeq: 12979699 ACK. v: SIP/2.0/UDP 192.168.16.91:5060. Max-Forwards: 70. Route: sip:7005874200@192.168.17.83;user=phone. User-Agent: MSTSYLVAIPGW. l: 0. .
U 2007/01/03 16:18:44.451221 192.168.16.192:5060 -> 192.168.17.83:5060 ACK sip:7005874200@192.168.17.83;user=phone SIP/2.0. Record-Route: sip:192.168.16.192;lr=on;ftag=d0b60bd7-1ffd6fd8-5b103598. t: sip:7005874200@siprt1.siptest.net:5060;user=phone;tag=6f6a15c679df8aa8. f:
sip:7006311229@192.168.16.91:5060;user=phone;tag=d0b60bd7-1ffd6fd8-5b10359
i: 1068bc1b-1bb-1ffd6fd8@192.168.16.91. CSeq: 12979699 ACK. Via: SIP/2.0/UDP 192.168.16.192;branch=z9hG4bK3f0e.9f290d35.2. v: SIP/2.0/UDP 192.168.16.91:5060. Max-Forwards: 69. User-Agent: MSTSYLVAIPGW. l: 0. .
U 2007/01/03 16:18:44.451328 192.168.16.91:5060 -> 192.168.16.192:5060 INVITE sip:192.168.16.192;lr=on;ftag=d0b60bd7-1ffd6fd8-5b103598 SIP/2.0. t: sip:7005874200@siprt1.siptest.net:5060;user=phone;tag=6f6a15c679df8aa8. f:
sip:7006311229@192.168.16.91:5060;user=phone;tag=d0b60bd7-1ffd6fd8-5b10359
Remote-Party-Id:
sip:7006311229@192.168.16.91:5060;user=phone;screen=yes;id-type=subscriber
;party=calling;privacy=off. Proxy-Require: privacy. i: 1068bc1b-1bb-1ffd6fd8@192.168.16.91. CSeq: 12979700 INVITE. v: SIP/2.0/UDP 192.168.16.91:5060. Max-Forwards: 70. Route: sip:7005874200@192.168.17.83;user=phone. m: sip:7006311229@192.168.16.91:5060;user=phone. c: application/sdp. Accept: application/sdp. Accept-Encoding: . Accept-Language: en. User-Agent: MSTSYLVAIPGW. l: 236. . v=0. o=MSTNT 536702936 536702937 IN IP4 192.168.16.91. s=Session SDP. c=IN IP4 192.168.16.91. t=0 0. m=audio 40878 RTP/AVP 18 101. a=silenceSupp:off. a=ecan:b on g168. a=ptime:20. a=rtpmap:101 telephone-event/8000. a=rtpmap:18 G729/8000.
U 2007/01/03 16:18:44.453398 192.168.16.192:5060 -> 192.168.16.91:5060 SIP/2.0 407 Proxy Authentication Required. t: sip:7005874200@siprt1.siptest.net:5060;user=phone;tag=6f6a15c679df8aa8. f:
sip:7006311229@192.168.16.91:5060;user=phone;tag=d0b60bd7-1ffd6fd8-5b10359
i: 1068bc1b-1bb-1ffd6fd8@192.168.16.91. CSeq: 12979700 INVITE. v: SIP/2.0/UDP 192.168.16.91:5060. Proxy-Authenticate: Digest realm="192.168.16.91", nonce="459c1ee0731e3291a5704b9666ffded6acb20bb5". Server: OpenSer (1.2.0-dev12-notls (i386/linux)). Content-Length: 0. Warning: 392 192.168.16.192:5060 "Noisy feedback tells: pid=14844 req_src_ip=192.168.16.91 req_src_port=5060 in_uri=sip:192.168.16.192;lr=on;ftag=d0b60bd7-1ffd6fd8-5b103598 out_uri=sip:7005874200@192.168.17.83;user=phone via_cnt==1". .
U 2007/01/03 16:18:44.471821 192.168.16.91:5060 -> 192.168.16.192:5060 ACK sip:192.168.16.192;lr=on;ftag=d0b60bd7-1ffd6fd8-5b103598 SIP/2.0. t: sip:7005874200@siprt1.siptest.net:5060;user=phone;tag=6f6a15c679df8aa8. f:
sip:7006311229@192.168.16.91:5060;user=phone;tag=d0b60bd7-1ffd6fd8-5b10359
i: 1068bc1b-1bb-1ffd6fd8@192.168.16.91. CSeq: 12979700 ACK. v: SIP/2.0/UDP 192.168.16.91:5060. Max-Forwards: 70. User-Agent: MSTSYLVAIPGW. l: 0. .
U 2007/01/03 16:18:44.472045 192.168.16.91:5060 -> 192.168.16.192:5060 BYE sip:192.168.16.192;lr=on;ftag=d0b60bd7-1ffd6fd8-5b103598 SIP/2.0. t: sip:7005874200@siprt1.siptest.net:5060;user=phone;tag=6f6a15c679df8aa8. f:
sip:7006311229@192.168.16.91:5060;user=phone;tag=d0b60bd7-1ffd6fd8-5b10359
i: 1068bc1b-1bb-1ffd6fd8@192.168.16.91. CSeq: 12979701 BYE. v: SIP/2.0/UDP 192.168.16.91:5060. Max-Forwards: 70. Route: sip:7005874200@192.168.17.83;user=phone. Accept: application/sdp. Accept-Encoding: . Accept-Language: en. User-Agent: MSTSYLVAIPGW. l: 0. .
U 2007/01/03 16:18:44.474251 192.168.16.192:5060 -> 192.168.17.83:5060 BYE sip:7005874200@192.168.17.83;user=phone SIP/2.0. Record-Route: sip:192.168.16.192;lr=on;ftag=d0b60bd7-1ffd6fd8-5b103598. t: sip:7005874200@siprt1.siptest.net:5060;user=phone;tag=6f6a15c679df8aa8. f:
sip:7006311229@192.168.16.91:5060;user=phone;tag=d0b60bd7-1ffd6fd8-5b10359
i: 1068bc1b-1bb-1ffd6fd8@192.168.16.91. CSeq: 12979701 BYE. Via: SIP/2.0/UDP 192.168.16.192;branch=z9hG4bKd29b.7f5f9c52.0. v: SIP/2.0/UDP 192.168.16.91:5060. Max-Forwards: 69. Accept: application/sdp. Accept-Encoding: . Accept-Language: en. User-Agent: MSTSYLVAIPGW. l: 0. .
U 2007/01/03 16:18:44.506433 192.168.17.83:5060 -> 192.168.16.192:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.16.192;branch=z9hG4bKd29b.7f5f9c52.0. Via: SIP/2.0/UDP 192.168.16.91:5060. Record-Route: sip:192.168.16.192;lr=on;ftag=d0b60bd7-1ffd6fd8-5b103598. From:
sip:7006311229@192.168.16.91:5060;user=phone;tag=d0b60bd7-1ffd6fd8-5b10359
To: sip:7005874200@siprt1.siptest.net:5060;user=phone;tag=6f6a15c679df8aa8. Call-ID: 1068bc1b-1bb-1ffd6fd8@192.168.16.91. CSeq: 12979701 BYE. User-Agent: Grandstream HT496 1.0.3.64 FXS0. Contact: sip:7005874200@192.168.17.83;user=phone. Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE. Supported: replaces. Content-Length: 0. .
U 2007/01/03 16:18:44.506549 192.168.16.192:5060 -> 192.168.16.91:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.16.91:5060. Record-Route: sip:192.168.16.192;lr=on;ftag=d0b60bd7-1ffd6fd8-5b103598. From:
sip:7006311229@192.168.16.91:5060;user=phone;tag=d0b60bd7-1ffd6fd8-5b10359
To: sip:7005874200@siprt1.siptest.net:5060;user=phone;tag=6f6a15c679df8aa8. Call-ID: 1068bc1b-1bb-1ffd6fd8@192.168.16.91. CSeq: 12979701 BYE. User-Agent: Grandstream HT496 1.0.3.64 FXS0. Contact: sip:7005874200@192.168.17.83;user=phone. Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE. Supported: replaces. Content-Length: 0. .
U 2007/01/03 16:18:47.652882 192.168.17.150:5060 -> 192.168.16.192:5060 ..................
-----Original Message----- From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] Sent: Wednesday, January 03, 2007 7:51 AM To: Shane Burrell Cc: users@openser.org Subject: Re: [Users] Issues with calls using openser.
The log shows that you challenge reINVITEs. MAybe this breaks the grandstream. Please try without challenging the reINVITE. If this helps, then it is a probably a grandstream bug.
But of course there is the question why the grandstream sends a reINVITE at all? This is often a codec problem.
Please also try to use other client (e.g. xlite) and post complete ngrep dumps: "ngrep -q -t -W byline port 5060"
regards klaus
Shane Burrell wrote:
The SIP UA was a grandstream ATA running the latest stable firmware.
Prior
to upgrading to 1.1 and moving to mediaproxy it worked well with the exception of good nat support which is why I would really like mediaproxy
to
work. Is there anything I should look for in the sip dialog to determine
if
the client, sip proxy, or the gateway is the culprit on disconnecting the call?
Thanks,
Shane
-----Original Message----- From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] Sent: Wednesday, January 03, 2007 4:52 AM To: Shane Burrell Cc: users@openser.org Subject: Re: [Users] Issues with calls using openser.
Maybe a bug in the caller's SIP client?
regards klaus
Shane Burrell wrote:
I recently installed the latest version of openser and this time used mediaproxy rather than rtpproxy. Everything seems to work but if a sip device is called the phone rings and is instantally disconnected and the
far
end is left off-hook. This worked before but I did modify my script to
work
with mediaproxy. Below is the wireshark decode of the sip messagining.
Any
help or suggestions on where to look would be great. Calls from the sip device works flawlessly. I am using a MaxTNT as the gateway.
|Time | 152.93.36.91 | siprt1.me.net| 152.93.37.83 |
|22.031 | INVITE SDP ( telephone-event) | |SIP From: sip:8385021101@192.168.16.91:5060 To:sip: 8385024200@ siprt1.me.net:5060
| |(5060) ------------------> (5060) | |
|22.031 | 100 Giving a try |
|SIP
Status
| |(5060) <------------------ (5060) | |
|22.031 | | INVITE SDP ( telephone-event) |SIP Request
| | |(5060) ------------------> (5060) |
|22.040 | | 100 Trying|
|SIP
Status
| | |(5060) <------------------ (5060) |
|22.042 | | 180 Ringing
|SIP
Status
| | |(5060) <------------------ (5060) |
|22.042 | 180 Ringing |
|SIP
Status
| |(5060) <------------------ (5060) | |
|25.244 | | 200 OK SDP ( telephone-event) |SIP Status
| | |(5060) <------------------ (5060) |
|25.245 | 200 OK SDP ( telephone-event) | |SIP Status
| |(5060) <------------------ (5060) | |
|25.269 | ACK | |
|SIP
Request
| |(5060) ------------------> (5060) | |
|25.269 | | ACK |
|SIP
Request
| | |(5060) ------------------> (5060) |
|25.269 | INVITE SDP ( telephone-event) | |SIP From: sip: 8385021101@152.93.36.91:5060 To:sip: 8385024200@ siprt1.me.net:5060
| |(5060) ------------------> (5060) | |
|25.270 | 407 Proxy Authentication Required | |SIP Status
| |(5060) <------------------ (5060) | |
|25.291 | ACK | |
|SIP
Request
| |(5060) ------------------> (5060) | |
|25.291 | BYE | |
|SIP
Request
| |(5060) ------------------> (5060) | |
|25.293 | | BYE |
|SIP
Request
| | |(5060) ------------------> (5060) |
|25.326 | | 200 OK |
|SIP
Status
| | |(5060) <------------------ (5060) |
|25.327 | 200 OK | |
|SIP
Status
| |(5060) <------------------ (5060) | |
Shane
Users mailing list Users@openser.org http://openser.org/cgi-bin/mailman/listinfo/users
-- Klaus Darilion nic.at
-
Klaus Darilion -
Shane Burrell