[asterisk-users] incoming call problem

[David fire]

sorry i cant help you :-(
i only can sugest add another peer in sip.conf in one use only audio and in
the other one use only T38.

you should post it again whit a subject like "T38 problem" or "please
help!!!!!! t38 problem".

David
Sorry again.



2009/2/27 michel freiha <michofr at gmail.com>

> Dear David,
>
> Please find on http://pastebin.com/m69b8559d my sip.conf file
>
> Thanks a lot
>
> On Fri, Feb 27, 2009 at 1:05 PM, David fire <ddfire at gmail.com> wrote:
>
>> paste your sip.conf.
>> David
>>
>> 2009/2/26 michel freiha <michofr at gmail.com>
>>
>>> Dear All,
>>> I have created an inbound context in SIP .conf that forward incoming call
>>> to opensips server...The problem appears as soon as I enable t38pt_udptl =
>>> yes...The Asterisk negotiate the SIP session with OpenSIPS without adding
>>> voice codec to INVITE packet...It just contains T.38 protocol...When
>>> t38pt_udptl is disabled everything looks OK and Ulaw is negotiated with
>>> OpenSIPS and cal success..Any suggestion here?
>>>
>>> Thanks
>>>
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>>
>>
>>
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>
>
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