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

[Brett Nemeroff]

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