I am running FreePBX 2.10.1.3 which is running on Asterisk 1.818.210.62-3. I have enabled T.38 passthru, added faxdetect=yes to the extension (going to Cisco ATA), added t38pt_udptl=yes under sip_general_custom.conf. I also added t38pt_udptl=yes to my outgoing settings on the SIP trunk to Flowroute. Flowroute has confirmed they support T.38 in both directions, yet I can’t get it to work to save my life. If choose Fax Passthru Method : ReINVITE on the ATA, I get the below error in the Asterisk log and the fax connection times out.
[2012-12-26 21:42:48] NOTICE[19807] chan_sip.c: FAX CNG detected but no fax extension
[2012-12-26 21:42:48] WARNING[19807] res_rtp_asterisk.c: Don’t know how to represent ‘f’
[2012-12-26 21:42:48] WARNING[19807] res_rtp_asterisk.c: Don’t know how to represent ‘f’
[2012-12-26 21:42:56] VERBOSE[19807] app_dial.c: – SIP/Flowroute-00000058 answered SIP/12502-00000057
[2012-12-26 21:42:57] VERBOSE[3455] netsock.c: == Using UDPTL TOS bits 184
[2012-12-26 21:42:57] VERBOSE[3455] netsock.c: == Using UDPTL CoS mark 5
[2012-12-26 21:42:57] NOTICE[3455] chan_sip.c: T.38 re-INVITE detected but no fax extension
[2012-12-26 21:42:57] VERBOSE[19807] netsock.c: == Using UDPTL TOS bits 184
[2012-12-26 21:42:57] VERBOSE[19807] netsock.c: == Using UDPTL CoS mark 5
[2012-12-26 21:42:57] WARNING[19807] res_rtp_asterisk.c: RTP Read too short
[2012-12-26 21:42:57] WARNING[19807] res_rtp_asterisk.c: RTP Read too short
[2012-12-26 21:42:57] WARNING[19807] res_rtp_asterisk.c: RTP Read too short
[2012-12-26 21:43:01] WARNING[19807] res_rtp_asterisk.c: RTP Read too short
If I change Fax Passthru Method to NSE, I get the below error and still times out. What gives? From my understanding, Asterisk 1.8 is supposed to support T.38 passthrough with no problems. Anyone have any ideas? Thanks for any help.
[2012-12-26 21:56:33] NOTICE[19986] chan_sip.c: FAX CNG detected but no fax extension
[2012-12-26 21:56:33] WARNING[19986] res_rtp_asterisk.c: Don’t know how to represent ‘f’
[2012-12-26 21:56:33] WARNING[19986] res_rtp_asterisk.c: Don’t know how to represent ‘f’
[2012-12-26 21:56:41] VERBOSE[19986] app_dial.c: – SIP/Flowroute-0000005c answered SIP/12502-0000005b
[2012-12-26 21:56:42] NOTICE[19986] res_rtp_asterisk.c: Unknown RTP codec 100 received from ‘192.168.2.100:16478’