On 06.01.2015 13:44, Frederic Van Espen wrote: > On Thu, Jan 1, 2015 at 7:09 PM, Recursive <li...@binarus.de> wrote: >> 1) Did anybody test T.38 with SPANDSP? If yes, which version of SPANDSP did >> you use? Mine is 0.0.6 PRE 20. Should I try to upgrade to PRE 21? Or to one >> of the snapshots? > > I remember that I was having issues with an older libspandsp library. > I simply upgraded to the latest available version which made the issue > go away. We were using 0.0.6 pre 12 before and upgraded to 0.0.6 pre > 20.
That's the version I am using. Nevertheless, I'll try to compile and use PRE 21 and the latest snapshot. I'll report the results. > However, I believe (and please someone do correct me if I'm wrong) > that libspandsp is not required when using t38 passthrough, which is > what you are trying to do. Then I'll just delete the SPANDSP module from the module directory (or compile a version without libspandsp it this does not work), test again and report the results. Could it make a difference if I use Digium's fax module instead of SPANDSP? > BTW, in the old thread I had asked if you saw anything in the logs > about the SIP message that was being retransmitted. Do you still see > retransmission with asterisk 13 and pjsip and a hangup after 30 > seconds? Currently I don't see such messages / timeouts, but this may be due to the fact that I currently don't get a T.38 connection which is lasting long enough. As far as I can remember, the 32 second timeout in my old chan_sip configuration startet exactly when the communication switched to T.38. Since PJSIP (or whoever) currently does not pass on the preamble to the local fax software, the switch to T.38 still happens, but the local fax software then says BYE after a few seconds (because obviously it does not detect the other end due to Asterisk not passing on the preamble packets). I'm not sure if the timeout would strike again if I had a T.38 "connection" for more than 5 (or so) seconds. >> 2) Recently, I have sent some questions about similar subjects to this list, >> and I have got helpful answers; people told me that I should *not* enable >> the fax *gateway* feature if both endpoints are capable of T.38. On the >> other hand, I have read (at multiple places) the the *gateway* code is >> responsible for detecting the v21 preamble. How does this fit together? > > AFAIK, when both ends support T.38, you don't need the gateway and can > turn to passthough mode. This is what I'm trying to do. Gateway is turned off in the dialplan and in the endpoints' description. > In passthrough mode, asterisk does not need > to detect anything. Just take the packet it receives and pass the > payload to the other end of the call. Well, that's what I have been thinking as well. But Asterisk does not pass on the packets which it receives from the ITSP's media gateway. Thank you very much, Recursive -- _____________________________________________________________________ -- 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
