search for: sip_answ

...ince I?d like to use g723 for sip communication between endpoints and * does not support it, I need to change codecs when a user wants to check voicemail, use a zap channel, etc. I have configured sip.conf and extensions.conf as below but when I try it I keep getting the following: chan_sip.c ...(sip_answer):Changing codec to GSM for this call because of ${SIP_CODEC} variable channel.c ..(ast_set_write_format): Unable to find a path from 2 to 1. Any ideas? sip.conf disallow=all allow=g723.1 allow=gsm extensions.conf exten => 1000,1,SetVar,SIP_CODEC=gsm exten => 1000,2,VoiceMailMain ___...

...1:55:33 DEBUG[1158913328]: File chan_sip.c, Line 2899 (build_route): build_route: Contact hop: 5010 <sip:5010@62.212.12.21> Jun 5 01:55:33 DEBUG[1236360496]: File pbx.c, Line 1116 (pbx_extension_helper): Launching 'Answer' Jun 5 01:55:33 DEBUG[1236360496]: File chan_sip.c, Line 934 (sip_answer): sip_answer(SIP/5010-d3c4) Jun 5 01:55:33 DEBUG[1236360496]: File pbx.c, Line 1116 (pbx_extension_helper): Launching 'BackGround' Jun 5 01:55:33 DEBUG[1236360496]: File channel.c, Line 1381 (ast_set_write_format): Set channel SIP/5010-d3c4 to write format 2 Jun 5 01:55:33 DEBUG[123636...

...041911234567@mypstngate|90") in new stack -- Called 0041911234567@mypstngate -- SIP/mypstngate-caed is making progress passing it to SIP/2016-b119 -- SIP/mypstngate-caed is ringing -- SIP/mypstngate-caed answered SIP/2016-b119 Jun 24 09:49:23 NOTICE[1094450096]: chan_sip.c:1314 sip_answer: Changing codec to 'g729' for this call because of ${SIP_CODEC) variable -- Attempting native bridge of SIP/2016-b119 and SIP/mypstngate-caed *CLI> sip show channels Peer User/ANR Call ID Seq (Tx/Rx) Lag Jitter Format 192.168.0.100 0041911234 1f7d34e...

...format ALAW Aug 26 15:54:17 DEBUG[-1260983376]: channel.c:1666 ast_set_write_format: Set channel SIP/2000-e12c to write format ALAW Aug 26 15:54:17 DEBUG[-1260983376]: channel.c:1699 ast_set_read_format: Set channel Zap/17-1 to read format ALAW Aug 26 15:54:17 DEBUG[-1260983376]: chan_sip.c:1824 sip_answer: sip_answer (SIP/2000-e12c) Aug 26 15:54:17 DEBUG[-1233335376]: chan_sip.c:817 __sip_ack: Stopping retransmission on '95AB5805-C94F-4C15-AC5A-6DFE5F58D644@192.168.1.101' of Response 46614: Found Aug 26 15:54:17 DEBUG[-1260983376]: rtp.c:378 ast_rtcp_read: Got RTCP report of 84 bytes Au...

...my SIP phone immediately hangs up. The other end keeps on ringing but when the callee answers, there is no sounds. I have found the "Didn't get frame from channel" error occurring in each such call. What does this mean? How can I fix it? -Mike- May 15 22:31:10 DEBUG[4792]: sip_answer(SIP/2433-9716) May 15 22:31:10 VERBOSE[4792]: -- Attempting native bridge of SIP/2433-9716 and SIP/2463-2f7a May 15 22:31:10 DEBUG[4792]: Got RTCP report of 84 bytes May 15 22:31:10 DEBUG[4792]: Ooh, format changed from unknown to ulaw May 15 22:31:10 DEBUG[4792]: Got RTCP report of 118 bytes...

Running asterisk 1.2.7 with latest zaptel on centos4.4. with Aastra 55i phones. Local outbound calling works fine, but ATT requires clients enter 7 digit code for long distance. All calls with 7 digit code are lost within 20 seconds of the call. This is the message I?m getting: Apr 19 12:38:16 WARNING[9615]: chan_sip.c:1228 retrans_pkt: Maximum retries exceeded on transmission

...th Festival 1.4.1. I've got a test extension setup, Festival configured and for some reason, when I dial that extension I get this: [Jan 10 17:16:05] WARNING[9082]: app_festival.c:511 festival_exec: Festival returned ER See the full debug below: [Jan 10 17:16:05] DEBUG[9082]: chan_sip.c:3381 sip_answer: SIP answering channel: SIP/SMEDIA-300-086da000 [Jan 10 17:16:05] DEBUG[9082]: chan_sip.c:6255 transmit_response_with_sdp: Setting framing from config on incoming call [Jan 10 17:16:05] DEBUG[9082]: chan_sip.c:6003 add_sdp: ** Our capability: 0x10c (ulaw|alaw|g729) Video flag: True [Jan 10 17:16:...

...128.91.56.38 [May 9 08:42:42] DEBUG[18512]: chan_sip.c:15336 sip_devicestate: Checking device state for peer 128.91.56.38 [May 9 08:42:42] DEBUG[18512]: channel.c:1026 channel_find_locked: Avoiding initial deadlock for channel '0xa29dd20' [May 9 08:42:42] DEBUG[20530]: chan_sip.c:3481 sip_answer: SIP answering channel: SIP/128.91.56.38-09c6e8f0 [May 9 08:42:42] DEBUG[20530]: chan_sip.c:6452 transmit_response_with_sdp: Setting framing from config on incoming call [May 9 08:42:42] DEBUG[20530]: chan_sip.c:6220 add_sdp: ** Our capability: 0x4 (ulaw) Video flag: True [May 9 08:42:42] D...

>> allow=ulaw >Why don't you remove this? Because I need some other users to be able to call out using ULAW over the same PSTN gateway... -Manuel ___________________________________________________ Ticinocom SA - Via Stazione 5 - 6600 Muralto Tel 0844 007070 - Fax 0844 007071 http://www.ticinocom.com

...p:192.168.20.1; answer-after=0") in new stack Oct 16 11:01:12 DEBUG[6767] pbx.c: Launching 'Page' Oct 16 11:01:12 VERBOSE[6767] logger.c: -- Executing Page("SIP/snom1-b7d0c328", "SIP/snom1&SIP/snom3|dq") in new stack Oct 16 11:01:12 DEBUG[6767] chan_sip.c: sip_answer(SIP/snom1-b7d0c328) Oct 16 11:01:12 DEBUG[6767] app_meetme.c: Building dynamic conference '2028709590d' Oct 16 11:01:12 DEBUG[6767] chan_zap.c: Using channel -2 Oct 16 11:01:12 VERBOSE[6767] logger.c: -- Created MeetMe conference 1023 for conference '2028709590d' Oct 16 11:0...

...rmat alaw Feb 13 23:50:54 DEBUG[28047] channel.c: Set channel Zap/1-1 to write format alaw Feb 13 23:50:54 DEBUG[28047] channel.c: Set channel Zap/1-1 to read format alaw Feb 13 23:50:54 DEBUG[28047] channel.c: Set channel SIP/46-62bb to write format alaw Feb 13 23:50:54 DEBUG[28047] chan_sip.c: sip_answer(SIP/46-62bb) Feb 13 23:50:54 DEBUG[27904] devicestate.c: Changing state for Zap/1 - state 2 (In use) Feb 13 23:50:54 DEBUG[27904] chan_sip.c: Checking device state for peer 46 Feb 13 23:50:54 DEBUG[27904] devicestate.c: Changing state for SIP/46 - state 2 (In use) Feb 13 23:50:54 DEBUG[28052] a...

Could anyone who has successfully configured Asterisk to use g729 to conference 10-20 people please post their configs. I purchased and successfully installed 2 g729 licenses and but when I dial into my conference number on the Asterisk box from a SPA-2000 set to allow all codecs, it always appears to connect using ULAW. My iax.conf file includes the following under the general section

...'Set' May 24 20:13:47 DEBUG[26803] pbx.c: Launching 'Goto' May 24 20:13:47 DEBUG[26803] pbx.c: Launching 'Set' May 24 20:13:47 DEBUG[26803] pbx.c: Launching 'Set' May 24 20:13:47 DEBUG[26803] pbx.c: Launching 'Answer' May 24 20:13:47 DEBUG[26803] chan_sip.c: sip_answer(SIP/XXX.XXX.XXX.XXX-b7b03730) May 24 20:13:47 DEBUG[26574] chan_sip.c: Checking device state for peer XXX.XXX.XXXX.XXX May 24 20:13:47 DEBUG[26574] channel.c: Avoiding initial deadlock for 'SIP/XXX.XXX.XXXX.XXX-b7b03730' May 24 20:13:47 DEBUG[26803] pbx.c: Launching 'Wait' May 2...

...'Set' May 24 20:13:47 DEBUG[26803] pbx.c: Launching 'Goto' May 24 20:13:47 DEBUG[26803] pbx.c: Launching 'Set' May 24 20:13:47 DEBUG[26803] pbx.c: Launching 'Set' May 24 20:13:47 DEBUG[26803] pbx.c: Launching 'Answer' May 24 20:13:47 DEBUG[26803] chan_sip.c: sip_answer(SIP/XXX.XXX.XXX.XXX-b7b03730) May 24 20:13:47 DEBUG[26574] chan_sip.c: Checking device state for peer XXX.XXX.XXXX.XXX May 24 20:13:47 DEBUG[26574] channel.c: Avoiding initial deadlock for 'SIP/XXX.XXX.XXXX.XXX-b7b03730' May 24 20:13:47 DEBUG[26803] pbx.c: Launching 'Wait' May 2...

...mat slin Mar 31 16:54:01 DEBUG[11747] channel.c: Set channel Zap/1-1 to write format slin Mar 31 16:54:01 DEBUG[11747] channel.c: Set channel Zap/1-1 to read format slin Mar 31 16:54:01 DEBUG[11747] channel.c: Set channel SIP/451-0e31 to write format slin Mar 31 16:54:01 DEBUG[11747] chan_sip.c: sip_answer(SIP/451-0e31) Mar 31 16:54:01 DEBUG[24349] devicestate.c: Changing state for Zap/1 - state 2 (In use) Mar 31 16:54:01 DEBUG[24349] chan_sip.c: Checking device state for peer 451 Mar 31 16:54:01 DEBUG[24349] devicestate.c: Changing state for SIP/451 - state 2 (In use) Mar 31 16:54:01 DEBUG[24349...

...xt=Exp_Netascom@172.25.96.48> Call-ID: 0ff188833a12041d6203999156019426@172.25.96.48 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Nov 21 14:17:48 VERBOSE[991] logger.c: -- SIP/evavox-08f09cb0 answered SIP/1649-08f029f0 Nov 21 14:17:48 DEBUG[991] chan_sip.c: sip_answer(SIP/1649-08f029f0) Nov 21 14:17:48 VERBOSE[991] logger.c: We're at 172.25.96.48 port 12058 Nov 21 14:17:48 VERBOSE[991] logger.c: Adding codec 0x100 (g729) to SDP Nov 21 14:17:48 VERBOSE[991] logger.c: Reliably Transmitting (no NAT) to 172.25.103.222:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 172....