----- Original Message ----- > Greetings-
> Working with the T.38 gateway functionality that is sparsely > documented [1], I'm attempting to get the following functional: > Asterisk calling system -> Asterisk system in T.38 Gateway Mode (box > in question) -> SIP Provider > The problem is: > -The provider is not initiating a reinvite to T.38, even though it is > 100% supported > -Asterisk is not detecting the CNG tones from the far side of the > call and initiating a T.38 session on that call leg (with the SIP > provider), but *DOES* attempt to initiate a T.38 session with the > calling Asterisk system (which rejects with SIP/488 as expected) > So, how does one force a reinvite to T.38 on the outbound call leg in > this scenario? I did find the same problem from another user on the > list in the archives, but didn't find a solution contained within > the responses [2]. > Thank you, > --Tim > [1] https://wiki.asterisk.org/wiki/display/AST/T.38+Fax+Gateway > [2] > http://lists.digium.com/pipermail/asterisk-users/2012-July/273535.html *bump* Any thoughts? I'm quite familiar with the T.38 functionality within Callweaver, and a function is provided there to do exactly what I need ( SipT38SwitchOver() ). However, given Callweaver is ancient at this point, and better T.38 features such as "gateway" do not function, I am pressed to use Asterisk (11.13.1) with SpanDSP (0.0.5, latest from Github since spandsp.org is down) for this job. :) Thanks! --Tim -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
