Hello all, we do have a problem here with Asterisk 11 talking T.38 to a t38modem 2.0. The callflow is:
ISDN PRI --> Berofix (10.1.1.150) --> Asterisk (10.1.1.148) --> t38modem (10.1.1.148) --> Hylafax [1] Although the call gets connected, both parties are unable to negotiate the audio codecs: [2013-01-23 21:59:57] VERBOSE[8805][C-00000000] chan_sip.c: Got T.38 offer in SDP in dialog [email protected]:5060 [2013-01-23 21:59:57] VERBOSE[8805][C-00000000] chan_sip.c: Capabilities: us - (alaw), peer - audio=(nothing)/video=(nothing)/text=(nothing), combined - (nothing) [2013-01-23 21:59:57] VERBOSE[8805][C-00000000] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing) [2013-01-23 21:59:57] VERBOSE[8805][C-00000000] chan_sip.c: Got T.38 Re-invite without audio. Keeping RTP active during T.38 session. which results in a "SIP/2.0 488 Not acceptable here" from the Asterisk and the call gets disconnected. Asterisk full log with SIP trace is at http://pastebin.com/NNr6BTdp This looks a lot like https://issues.asterisk.org/jira/browse/ASTERISK-15596 which doesn't seems to be solved by now. Is this still a known issue in Asterisk 11? What can I do to make Asterisk 11 play nice with t38modem v2.0? Environment: -------------- Red Hat Enterprise Linux Server release 6.3 (Santiago) Linux myhost.mydomain.local 2.6.32-279.19.1.el6.x86_64 #1 SMP Sat Nov 24 14:35:28 EST 2012 x86_64 x86_64 x86_64 GNU/Linux T38Modem Version 2.0.0 (OPAL-3.9.0/3.9beta0, PTLIB-2.9.0/2.9beta0 (svn:24165)) by Vyacheslav Frolov on Unix Linux (2.6.32-279.19.1.el6.x86_64-x86_64) Asterisk 11.1.0 Hylafax 6.0.6 Thanx in advance and greetings, Carsten. [1] Yes, I know: the Berofix appliance can talk directly to the t38modems, which works perfectly well here. But there is a limitation of 140 SIP Accounts in the Berofix and we have to serve ~500 fax numbers. So we had to set Asterisk between the Berofix and the t38modems, bearing the SIP accounts. -- Blinkenlichten (Maass & Sacha GbR) - Open Source Solutions Weigandufer 45 - 12059 Berlin - http://www.blinkenlichten.de FON: ++49 +30 13896247 - MAIL: [email protected] FAX: ++49 +30 13896249 - PGP: Key Id 0x2CBCA806 St.Nr. 16/274/61636 -- _____________________________________________________________________ -- 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
