If this is possible at all, I would suspect it might require a rather creative home-grown scenario in the B2BUA module. Since the call is already established, a simple proxy isn't going to be able to tear down the one side and reestablish it with another endpoint.
- Jeff On Mar 17, 2010, at 11:43 AM, 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 > > -- > Matthew S. Crocker > President > Crocker Communications, Inc. > PO BOX 710 > Greenfield, MA 01302-0710 > http://www.crocker.com > P: 413-746-2760 > > > _______________________________________________ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
