[Users] Issues with calls using openser.
Shane Burrell
shaneb at metrostat.net
Wed Jan 3 13:41:14 CET 2007
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 at pernau.at]
Sent: Wednesday, January 03, 2007 4:52 AM
To: Shane Burrell
Cc: users at 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 at 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 at 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
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