Hi Jeff, as opensips will act as b2b, your call will be actually split in 2 calls (from SIP point of view) - a call C1 from GW to opensips and another one C2 from opensips to UAC. So at re-INVITE time, opensips b2b will hung up C2 and replace it with a C3 to a new destination, bridging it with C1
Regards, Bogdan Jeff Kronlage wrote: > I'm confused on this as well - wouldn't you be effectively placing two > calls (one via a non-T38 gateway, one via a T38 gateway) to the same > destination? Figuring that most T38 is going to terminate to a single > analog device, I would think that were this possible at a SIP level, the > device would already be "busy" before the second call came in as fax > machines don't typically drop the line very rapidly? > > Jeff > > -----Original Message----- > From: [email protected] > [mailto:[email protected]] On Behalf Of Bogdan-Andrei > Iancu > Sent: Wednesday, March 17, 2010 11:23 AM > To: OpenSIPS users mailling list > Subject: Re: [OpenSIPS-Users] T.38 detection/redirect in OpenSIPS > > 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 >>>>> >>>>> >>>>> -- Bogdan-Andrei Iancu www.voice-system.ro _______________________________________________ Users mailing list [email protected] http://lists.opensips.org/cgi-bin/mailman/listinfo/users
