Are you trying to send fax over T.38? As far I understand * don't support T.38 event when passing packets trouth. I'm interested in T.38 support too, so if anybody could explain why * can't just pass theese packets (as i undrstand there is no need foe recoding etc.) I would be very appreciative. Are anybody currently working on T.38 support for * ? I don't mean T.38 support on zap interfaces, just passing T.38 packets trouth asterisk
-- Sincerely, Elman Efendiyev [EMAIL PROTECTED] -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kai Militzer Sent: Tuesday, November 23, 2004 6:04 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Fax over SIP Problems (sorry for this topic ...) Hello everyone! I tried to send a fax over SIP with an Asterisk Server in the middle (no Digium Cards, etc. installed, everything is SIP only, the PSTN-Gateway is external). Whenever I start sending a Fax to a PSTN destination, the Call gets answered and asterisk tries to build a native bridging: -- Attempting native bridge of SIP/sip.westend.com-082fd1b8 and SIP/xxx-3ef8 Then the following messages start to appear (about 100 of them) Nov 23 16:27:25 NOTICE[1061908]: rtp.c:489 ast_rtp_read: Unknown RTP codec 127 received After that asterisk gets totaly confused: Nov 23 16:27:33 WARNING[1061908]: rtp.c:428 ast_rtp_read: RTP Read too short Nov 23 16:27:33 WARNING[1061908]: rtp.c:428 ast_rtp_read: RTP Read too short Nov 23 16:27:33 WARNING[1061908]: rtp.c:428 ast_rtp_read: RTP Read too short Nov 23 16:27:33 WARNING[1061908]: rtp.c:428 ast_rtp_read: RTP Read too short Nov 23 16:27:35 NOTICE[1061908]: rtp.c:489 ast_rtp_read: Unknown RTP codec 1 received Nov 23 16:27:35 WARNING[1061908]: codec_gsm.c:135 gsmtolin_framein: Huh? A GSM frame that isn't a multiple of 33 or 65 bytes long from RTP (15)? Nov 23 16:27:35 NOTICE[1061908]: channel.c:1724 ast_set_read_format: Unable to find a path from G723 to ALAW Nov 23 16:27:35 NOTICE[1061908]: channel.c:1691 ast_set_write_format: Unable to find a path from GSM to G723 Nov 23 16:27:35 WARNING[1061908]: codec_gsm.c:135 gsmtolin_framein: Huh? A GSM frame that isn't a multiple of 33 or 65 bytes long from RTP (20)? Nov 23 16:27:35 WARNING[1061908]: channel.c:2115 ast_channel_make_compatible: No path to translate from SIP/sip.westend.com-082fd1b8(8) to SIP/xxx-3ef8(1) Nov 23 16:27:35 WARNING[1061908]: channel.c:2633 ast_channel_bridge: Can't make SIP/sip.westend.com-082fd1b8 and SIP/xxx-3ef8 compatible Nov 23 16:27:35 WARNING[1061908]: res_features.c:358 ast_bridge_call: Bridge failed on channels SIP/sip.westend.com-082fd1b8 and SIP/xxx-3ef8 == Spawn extension (macro-enumcall, s, 211) exited non-zero on 'SIP/sip.westend.com-082fd1b8' in macro 'enumcall' == Spawn extension (xxx, 911879, 7) exited non-zero on 'SIP/sip.westend.com-082fd1b8' -- Executing NoOp("SIP/sip.westend.com-082fd1b8", "") in new stack > cdr_odbc: Query Successful! Then the call gets hung up. I cannot explain why this happens. I would have explanations for the fax-machines not able to synchronize or faxes not being transmitted correctly, as the communication is SIP only, but this seems a bit strange to me. I can't even tell, if my asterisk produces these messages, or the other side (aka PSTN-Gateway). If anyone can bring some light into this behavior I would be very greatful. Best regards Kai -- Kai Militzer WESTEND GmbH | Internet-Business-Provider Technik CISCO Systems Partner - Authorized Reseller Lьtticher StraЯe 10 Tel 0241/701333-11 [EMAIL PROTECTED] D-52064 Aachen Fax 0241/911879 _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
