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
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