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
_______________________________________________
Users mailing list
[email protected]
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
