[OpenSIPS-Users] T.38 detection/redirect in OpenSIPS

[Bogdan-Andrei Iancu]

right, that is exactly what the b2b is up to do - to be able (at 
signalling level) to manipulate the call legs

Regards,
Bogdan

Brett Nemeroff wrote:
> Bogdan,
> But at this point, you are now playing with a dialg that is already
> connected to an endpoint. You'd need to drop the first call to
> establish a new call with the reinvite. Right?
> -Brett
>
> On Mar 17, 2010, at 11:50 AM, Bogdan-Andrei Iancu <bogdan at voice-system.ro
>  > wrote:
>
>   
>> Hi Brett,
>>
>> Brett Nemeroff wrote:
>>     
>>> I don't think there is any way to do this without an RTP capable
>>> device in the mix.
>>>       
>> you do not need to look into RTP as the FAX is advertised in the
>> re-INVITE (in SDP) - so you can detect it from opensips script by
>> inspecting the SDP of reINVITES
>>     
>>> What you may be able to do is have asterisk detect that it's a fax,
>>> then reject it if it is.. I don't know if you can do all that without
>>> answering the call.
>>>       
>> no, you cannnot, as first the call is established (from sip point of
>> view) as a simple audio call and after that re-negotiated (via
>> re-INVITE) for FAX
>>     
>>> Then you can forward it back to the proxy if it is a fax with maybe a
>>> prefix.
>>>
>>> A lot of assumptions in there. Would like to hear if you find
>>> something that works. Not sure if you can SIP Spiral yet in asterisk
>>> anyway. ;)
>>>       
>> I do not see the need of Asterisk - maybe with some changes, the b2b
>> module will be able to handle this - see my prev email.
>>
>> Regards,
>> Bogdan
>>
>>     
>>> -Brett
>>>
>>>
>>> On Wed, Mar 17, 2010 at 10:51 AM, David J. <david at styleflare.com
>>> <mailto:david at styleflare.com>> wrote:
>>>
>>>    Matt,
>>>
>>>    I am for sure probably wrong, but I think you would need
>>> Asterisk or
>>>    Variant to Determine that it is a Fax Call,
>>>    I dont think UAC's send T38 information without negotiating with
>>> the
>>>    other side who request that it is capable, then it brings you to
>>>    Jeff's
>>>    answer.
>>>
>>>    See above.
>>>
>>>
>>>    Matthew S. Crocker wrote:
>>>       
>>>> Can OpenSIPS make routing decisions based on the SDP information
>>>>         
>>>    in an INVITE?
>>>       
>>>> Lets say I have the following config
>>>>
>>>> PSTN -> t.38 Gateway -> OpenSIPS ->  UserAgent
>>>>
>>>> I have a TN from the PSTN routed to the UserAgent,  I'd like to
>>>>         
>>>    provide a service so the user can use the TN for both voice &
>>> faxing.
>>>       
>>>> Voice call goes through normally (g.711 g.729 codec)
>>>>
>>>> Fax call starts off as a normal voice call (INVITE, 180, 183,
>>>>         
>>>    200).  Once the call is answered the originating end (PSTN) starts
>>>    sending fax tones. The Gateway hears the fax tones and attempts to
>>>    RE-INVITE with T.38 in the SDP.  I'd like OpenSIPS to see the T.38
>>>    capability in the SDP and redirect the call to a fax->e-mail
>>>    gateway.  So,  the 2nd INVITE comes in, OpenSIPS sends the INVITE
>>>    to the fax gateway and a BYE to the user.  The fax gateway does a
>>>    200 and negotiates T.38 with the PSTN gateway.
>>>       
>>>> I know I can route the call through Asterisk and have it do a
>>>>         
>>>    quiet answer and listen for the modem sounds.  I'd like to avoid
>>>    using Asterisk for all RTP traffic and only use it for the fax
>>>    gateway traffic (i.e. once it has been determined to be a fax
>>>    Asterisk steps in and handled the T38 -> E-mail)
>>>       
>>>> -Matt
>>>>
>>>>
>>>>         
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>>>
>>>
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>>>
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>>>       
>> --
>> Bogdan-Andrei Iancu
>> www.voice-system.ro
>>
>>
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>>     
>
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>   


-- 
Bogdan-Andrei Iancu
www.voice-system.ro