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

[Bogdan-Andrei Iancu]

Hi Matthew,

you do not need to look into the media part, as you can "spot" the FAX 
presence via the re-INVITE with T38 codec in SDP (you can detect it from 
opensips cfg).

So, maybe using the b2b module for something like:
    - allow the voice call to be setup via the b2b in a transparent way
    - if the re-INVITE wth T38 is received from GW, b2b will close the 
leg to the users UA and create a new leg to something able to handle the 
fax - of course, the b2b will bridge the existing leg (towards PSTN) and 
the new leg.

Regards,
Bogdan


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
>
>   


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