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 <bog...@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. <da...@styleflare.com >> <mailto:da...@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 >>> >>> >> >> >> _______________________________________________ >> Users mailing list >> Users@lists.opensips.org <mailto: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 >> > > > -- > Bogdan-Andrei Iancu > www.voice-system.ro > > > _______________________________________________ > 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