@ All, This has very exciting possibilities for me. I'm definitely going to look at the b2b module and test this out. I will share my findings.
regards On Thu, Mar 18, 2010 at 10:32 AM, Bogdan-Andrei Iancu < [email protected]> wrote: > 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 > -- TC
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