3 Jan
2007
3 Jan
'07
7:41 a.m.
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(a)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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--
Klaus Darilion
nic.at