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. <[email protected] > <mailto:[email protected]>> 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 > > > > > > > _______________________________________________ > Users mailing list > [email protected] <mailto:[email protected]> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > ------------------------------------------------------------------------ > > _______________________________________________ > Users mailing list > [email protected] > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- Bogdan-Andrei Iancu www.voice-system.ro _______________________________________________ Users mailing list [email protected] http://lists.opensips.org/cgi-bin/mailman/listinfo/users
