[Users] Issues with calls using openser.
Klaus Darilion
klaus.mailinglists at pernau.at
Wed Jan 3 13:51:05 CET 2007
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 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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>> Users mailing list
>> Users at openser.org
>> http://openser.org/cgi-bin/mailman/listinfo/users
>
>
--
Klaus Darilion
nic.at
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