search for: __ast_read

....931 (8) len=5 > Call Ref: len= 2 (reference 499/0x1F3) (Originator) > Message type: CONNECT ACKNOWLEDGE (15) [Oct 23 10:39:56] DEBUG[6136]: chan_zap.c:1413 zt_enable_ec: Echo cancellation already on -- Zap/3-1 answered SIP/test.com-08dc1ef8 [Oct 23 10:39:58] DTMF[13914]: channel.c:2346 __ast_read: DTMF end '5' received on SIP/test.com-08dc1ef8, duration 0 ms [Oct 23 10:39:58] DTMF[13914]: channel.c :2382 __ast_read: DTMF begin emulation of '5' with duration 100 queued on SIP/test.com-08dc1ef8 [Oct 23 10:39:58] DEBUG[13914]: chan_zap.c:1048 zt_digit_begin: Started VLDTMF digi...

...le log from a failing call (non-significant numbers obscured to protect the guilty!). *CLI> -- Accepting overlap call from '1780471800' to '0173' on channel 0/15, span 2 -- Starting simple switch on 'Zap/46-1' [Jun 28 12:00:32] DTMF[6387]: channel.c:2297 __ast_read: DTMF end '3' received on Zap/46-1, duration 0 ms [Jun 28 12:00:32] DTMF[6387]: channel.c:2297 __ast_read: DTMF end '2' received on Zap/46-1, duration 0 ms [Jun 28 12:00:33] DTMF[6387]: channel.c:2297 __ast_read: DTMF end '3' received on Zap/46-1, duration 0 ms [Jun 28 12...

Is there an issue with running it with the latest from the 1.6.2 branch? I did an svn update and make install and now when somebody comes in via Skype, I get an infinite loop of: [Apr 27 09:53:29] WARNING[10471]: channel.c:2701 __ast_read: read() failed: Invalid argument [Apr 27 09:53:29] WARNING[10471]: channel.c:2701 __ast_read: read() failed: Invalid argument [Apr 27 09:53:29] WARNING[10471]: channel.c:2701 __ast_read: read() failed: Invalid argument [Apr 27 09:53:29] WARNING[10471]: channel.c:2701 __ast_read: read() failed: Inva...

...ude => parkedcalls [macro-parkswitch] exten => s,1,ParkAndAnnounce(,,PARKED,SIP/100) messages: [Jan 30 21:00:00] DEBUG[7114][C-00000000]: res_rtp_asterisk.c:2847 create_dtmf_frame: Creating BEGIN DTMF Frame: 42 (*), at x.x.x.x:9530 [Jan 30 21:00:00] DTMF[7114][C-00000000]: channel.c:4050 __ast_read: DTMF begin '*' received on SIP/at-tcty-ssw-00000000 [Jan 30 21:00:00] DTMF[7114][C-00000000]: channel.c:4061 __ast_read: DTMF begin passthrough '*' on SIP/at-tcty-ssw-00000000 [Jan 30 21:00:00] DEBUG[7114][C-00000000]: res_rtp_asterisk.c:2165 ast_rtp_update_source: Setting the m...

All, Im having a problem with ReceiveFax where its generating a ton of these messages the entire time the receivefax app is running receiving my fax. [Apr 7 22:16:06] ERROR[26918]: channel.c:2520 __ast_read: ast_read() called with no recorded file descriptor. Im running on Centos 5.2 with all patches. asterisk-1.6.0.9 asterisk-addons-1.6.0.1 dahdi-linux-2.1.0.4 dahdi-tools-2.1.0.2 libpri-1.4.9 spandsp-0.0.5pre4 The receivefax app works perfectly, ie i am able to receive the faxes, and what not, but...

...fbridge.conf': dtmf_passthrough=yes From logger.conf, I can see the DTMF tones via setting "console => dtmf". When I dial into the conference bridge with a SIP UA and dial 9, for example, this is what I see: sip1*CLI> [Dec 19 01:29:50] DTMF[22561][C-000005ba]: channel.c:4164 __ast_read: DTMF begin '9' received on SIP/3002-0000003d [Dec 19 01:29:50] DTMF[22561][C-000005ba]: channel.c:4175 __ast_read: DTMF begin passthrough '9' on SIP/3002-0000003d [Dec 19 01:29:50] DTMF[22561][C-000005ba]: channel.c:4078 __ast_read: DTMF end '9' received on SIP/3002-0000003...

My last problem was nicely solved through this mailing list so hopefully this new problem will have the same happy outcome. My situation is that I have many extensions. Here is a sample: [client-phone](!) type=friend host=dynamic secret=XXXXXXXXXX dtmfmode=auto disallow=all allow=ulaw allow=gsm allow=g723 allow=ilbc subscribemwi=no [4165555555](client-phone) secret=xxxxxxxxxxxxxxxxxxxxxxxxxx

...packet 8, (1, 42) Tx-Frame Retry[-01] -- OSeqno: 012 ISeqno: 009 Type: IAX Subclass: ACK Timestamp: 27464ms SCall: 00002 DCall: 08434 [127.0.0.1:40310] [Jul 7 14:41:11] DEBUG[1261]: chan_iax2.c:8273 socket_process: For call=2, set last=27464 [Jul 7 14:41:11] DTMF[9852]: channel.c:2400 __ast_read: DTMF end '*' received on IAX2/6001-2, duration 0 ms [Jul 7 14:41:11] DTMF[9852]: channel.c:2436 __ast_read: DTMF begin emulation of '*' with duration 100 queued on IAX2/6001-2 [Jul 7 14:41:11] DEBUG[9852]: channel.c:4163 ast_generic_bridge: Got DTMF begin on channel (IAX2/6001...

[image: Post]<http://accessanywebsite.com/search.php?u=Oi8vd3d3LmVmbG8ubmV0L1ZJQ0lESUFMZm9ydW0vdmlld3RvcGljLnBocD9wPTI4NjU%3D&b=2#28652>Posted: Thu Apr 09, 2009 8:34 pm Post subject: DTMF and IVR ... Sorry but URGENT[image: Reply with quote]<http://accessanywebsite.com/search.php?u=Oi8vd3d3LmVmbG8ubmV0L1ZJQ0lESUFMZm9ydW0vcG9zdGluZy5waHA%2FbW9kZT1xdW90ZSZwPTI4NjUy&b=2>

...BUG[29533]: rtp.c:1148 ast_rtcp_read: Got RTCP report of 60 bytes [Oct 2 11:09:20] DEBUG[29533]: rtp.c:958 process_rfc2833: - RTP 2833 Event: 0000000b (len = 4) [Oct 2 11:09:20] DEBUG[29533]: rtp.c:806 send_dtmf: Sending dtmf: 35 (#), at 192.168.0.148 [Oct 2 11:09:20] DTMF[29533]: channel.c:2840 __ast_read: DTMF begin '#' received on SIP/1000-0a292360 [Oct 2 11:09:20] DTMF[29533]: channel.c:2850 __ast_read: DTMF begin passthrough '#' on SIP/1000-0a292360 [Oct 2 11:09:20] DEBUG[29533]: channel.c:4806 ast_generic_bridge: Got DTMF begin on channel (SIP/1000-0a292360) [Oct 2 11:09:20]...

...BUG[29533]: rtp.c:1148 ast_rtcp_read: Got RTCP report of 60 bytes [Oct 2 11:09:20] DEBUG[29533]: rtp.c:958 process_rfc2833: - RTP 2833 Event: 0000000b (len = 4) [Oct 2 11:09:20] DEBUG[29533]: rtp.c:806 send_dtmf: Sending dtmf: 35 (#), at 192.168.0.148 [Oct 2 11:09:20] DTMF[29533]: channel.c:2840 __ast_read: DTMF begin '#' received on SIP/1000-0a292360 [Oct 2 11:09:20] DTMF[29533]: channel.c:2850 __ast_read: DTMF begin passthrough '#' on SIP/1000-0a292360 [Oct 2 11:09:20] DEBUG[29533]: channel.c:4806 ast_generic_bridge: Got DTMF begin on channel (SIP/1000-0a292360) [Oct 2 11:09:20]...

I am having difficulties with customer-bound DTMF being very short & clipped off (and basically unusable, as systems on the customer side aren't recognizing the DTMF digits, and I can barely tell that DTMF is there when I listen on a handset). My system set up as follows: PSTN <--> Metaswitch <-SIP-> Asterisk <-SIP or IAX2-> CPE Asterisk is running Asterisk 10.4.0 on a

...BUG[29533]: rtp.c:1148 ast_rtcp_read: Got RTCP report of 60 bytes [Oct 2 11:09:20] DEBUG[29533]: rtp.c:958 process_rfc2833: - RTP 2833 Event: 0000000b (len = 4) [Oct 2 11:09:20] DEBUG[29533]: rtp.c:806 send_dtmf: Sending dtmf: 35 (#), at 192.168.0.148 [Oct 2 11:09:20] DTMF[29533]: channel.c:2840 __ast_read: DTMF begin '#' received on SIP/1000-0a292360 [Oct 2 11:09:20] DTMF[29533]: channel.c:2850 __ast_read: DTMF begin passthrough '#' on SIP/1000-0a292360 [Oct 2 11:09:20] DEBUG[29533]: channel.c:4806 ast_generic_bridge: Got DTMF begin on channel (SIP/1000-0a292360) [Oct 2 11:09:20]...

...BUG[29533]: rtp.c:1148 ast_rtcp_read: Got RTCP report of 60 bytes [Oct 2 11:09:20] DEBUG[29533]: rtp.c:958 process_rfc2833: - RTP 2833 Event: 0000000b (len = 4) [Oct 2 11:09:20] DEBUG[29533]: rtp.c:806 send_dtmf: Sending dtmf: 35 (#), at 192.168.0.148 [Oct 2 11:09:20] DTMF[29533]: channel.c:2840 __ast_read: DTMF begin '#' received on SIP/1000-0a292360 [Oct 2 11:09:20] DTMF[29533]: channel.c:2850 __ast_read: DTMF begin passthrough '#' on SIP/1000-0a292360 [Oct 2 11:09:20] DEBUG[29533]: channel.c:4806 ast_generic_bridge: Got DTMF begin on channel (SIP/1000-0a292360) [Oct 2 11:09:20]...

...facing the same problem and did not see one single answer to his post. Maybe I'll be a little bit more lucky. When I'm trying to issue an Atxfer AMI command, in the asterisk 1.8 branch, what happens is that some DTMF's are sent, like this : [Mar 22 15:46:27] DTMF[5910]: channel.c:3900 __ast_read: DTMF end '1' received on SIP/bmwgsjrponciuj-0000009f, duration 0 ms [Mar 22 15:46:27] DTMF[5910]: channel.c:3926 __ast_read: DTMF begin emulation of '1' with duration 100 queued on SIP/bmwgsjrponciuj-0000009f [Mar 22 15:46:27] DTMF[5910]: channel.c:4018 __ast_read: DTMF end emulati...

...0000175> Playing '/var/spool/asterisk/voicemail/VoiceMail/stocktrans2/unavail.gsm' (language 'en')<<< > 0x7f7fea5dc000 -- Probation passed - setting RTP source address to 72.143.94.110:28503<<< [Apr 18 11:45:38] DTMF[-1][C-0004c47b]: channel.c:4215 __ast_read: DTMF begin '*' received on SIP/alex-00000175<<< [Apr 18 11:45:38] DTMF[-1][C-0004c47b]: channel.c:4219 __ast_read: DTMF begin ignored '*' on SIP/alex-00000175<<< [Apr 18 11:45:38] DTMF[-1][C-0004c47b]: channel.c:4129 __ast_read: DTMF end '*' received on...

Hi list! ConfBridge dtmf_passthrough=no doesn't seem to have any effect. DTMF gets transmitted throughout the conference. I've tried Asterisk 10.7.1 from the official RPMs and 10.8.0 compiled from source. I've confirmed that it's disabled via the CLI "confbridge show profile user <profilename>". It's an all-SIP scenario with RFC2833 as the DTMF protocol.

Hello, I installed Asterisk 1.6.2.17.3 ( latest as of yesterday ) and had multiple problems with DTMF. I have two machines, we'll call them asterisk and asterisk-pri. Asterisk does IVR and asterisk-pri has a PRI card in it and connects to the PSTN. The two servers communicate via SIP with RFC2833. I setup logger.conf on both machines to display DTMF to the console. Both are built from

...oice-dialout:3] AGI("DAHDI/18-1", "smvoice) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/smvoice -- Playing '/home/silentm/record/please_press/one_to_call.' (escape_digits=0123456789*#) (sample_offset 0) [May 3 21:47:38] DTMF[21746]: channel.c:2368 __ast_read: DTMF end '1' received on DAHDI/18-1, duration 0 ms [May 3 21:47:38] DTMF[21746]: channel.c:2423 __ast_read: DTMF end accepted without begin '1' on DAHDI/18-1 [May 3 21:47:38] DTMF[21746]: channel.c:2434 __ast_read: DTMF end passthrough '1' on DAHDI/18-1 -- Playing...

...email(${DID_NO}) exten => 0,1,Voicemail(${DID_NO}) exten => a,1,VoiceMailMain(${DID_NO}) exten => h,1,Hangup In version 1.2, when I hit "0" during the playback, I will be directed to voicemail. But in verison 1.4, the call hangs up. [Jan 24 16:05:37] DTMF[5754]: channel.c:2148 __ast_read: DTMF begin '0' received on SIP/5060-08c53e68 [Jan 24 16:05:37] DTMF[5754]: channel.c:2128 __ast_read: DTMF end '0' received on SIP/5060-08c53e68 == Spawn extension (play_recording, s, 1) exited non-zero on 'SIP/5060-08c53e68' -- Executing [h@play_recording:1] Hangup(&quo...