right, that is exactly what the b2b is up to do - to be able (at signalling level) to manipulate the call legs
Regards, Bogdan Brett Nemeroff wrote: > 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 <[email protected] > > 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. <[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 >> > > _______________________________________________ > 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
