My 2 cents worth. Turn off faxdetect in the peer configuration for Asterisk.
Failing that, try the Fax Gateway feature in Asterisk 11 to Hylafax listening on an IAX2 channel.
Larry. On 24/01/2013 6:32 AM, Carsten Maass wrote:
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.
-- _____________________________________________________________________ -- 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
