search for: ast_settimeout

...Line 3428 (zt_read): DTMF digit: 6 on Zap/18-1 Oct 6 10:55:13 DEBUG[27664]: File chan_zap.c, Line 3428 (zt_read): DTMF digit: 9 on Zap/18-1 Oct 6 10:55:13 DEBUG[27664]: File chan_zap.c, Line 1056 (zt_enable_ec): No echocancellation requested Oct 6 10:55:16 DEBUG[27664]: File channel.c, Line 953 (ast_settimeout): Scheduling timer at 160 sample intervals Oct 6 10:55:16 DEBUG[27664]: File chan_zap.c, Line 3428 (zt_read): DTMF digit: 7 on Zap/18-1 Oct 6 10:55:21 DEBUG[27664]: File channel.c, Line 953 (ast_settimeout): Scheduling timer at 0 sample intervals Oct 6 10:55:21 DEBUG[27664]: File channel.c, Line...

...ING[6133]: channel.c:1901 ast_request: No channel type registered for 'zap' Jan 27 11:02:23 WARNING[6133]: app_meetme.c:227 build_conf: Unable to open pseudo channel - trying device -- Created MeetMe conference 1023 for conference '81' Jan 27 11:02:23 DEBUG[6133]: channel.c:1128 ast_settimeout: Scheduling timer at 160 sample intervals -- Playing 'conf-onlyperson' (language 'si') Jan 27 11:02:23 DEBUG[6133]: chan_iax2.c:5346 socket_read: Ooh, voice format changed to 2 Jan 27 11:02:27 DEBUG[6133]: channel.c:1128 ast_settimeout: Scheduling timer at 0 sample intervals Jan...

...Line 3249 (build_route): build_route: Contact hop : <sip:192.168.7.3> -- Executing VoiceMailMain2("SIP/xirak-259d", "") in new stack DEBUG[294927]: File rtp.c, Line 1007 (ast_rtp_write): Ooh, format changed from U NKN to ULAW DEBUG[294927]: File channel.c, Line 947 (ast_settimeout): Scheduling timer at 16 0 sample intervals -- Playing 'vm-login' DEBUG[81926]: File chan_sip.c, Line 540 (__sip_ack): Stopping retransmission on '6E5D898E-492D-400B-A42B-8B25FE25F2EE@192.168.7.3' of Response 1: Found DEBUG[294927]: File channel.c, Line 947 (ast_settimeout): S...

...channel SIP/gafachi1a-00000000 to write format slin [Aug 4 17:50:11] DEBUG[17094]: rtp.c:3878 ast_rtp_write: Ooh, format changed from unknown to ulaw [Aug 4 17:50:11] DEBUG[17094]: rtp.c:3904 ast_rtp_write: Created smoother: format: 4 ms: 20 len: 160 [Aug 4 17:50:11] DEBUG[17094]: channel.c:2488 ast_settimeout: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Aug 4 17:50:11] -- <SIP/gafachi1a-00000000> Playing 'digits/1.slin' (language 'en') [Aug 4 17:50:12] DEBUG[17094]: channel.c:2488 ast_settimeout: Scheduling timer at (571 requested / 100 actual) time...

...xecuting Wait("SIP/sip3-768a", "1") in new stack -- Executing AgentLogin("SIP/sip3-768a", "") in new stack Aug 12 16:31:37 DEBUG[1127562160]: rtp.c:1156 ast_rtp_write: Ooh, format changed from UNKN to ULAW Aug 12 16:31:37 DEBUG[1127562160]: channel.c:1101 ast_settimeout: Scheduling timer at 160 sample intervals -- Playing 'agent-user' (language 'en') Aug 12 16:31:37 DEBUG[1103408048]: chan_sip.c:817 __sip_ack: Stopping retransmission on '78383678327d335d' of Response 2: Found Aug 12 16:31:41 DEBUG[1127562160]: channel.c:1101 ast_settime...

...ecuting DISA("Zap/66-1", "no-password| mitel-chantilly") in new stack Nov 30 02:44:54 DEBUG[4684]: app_disa.c:160 disa_exec: Context: mitel-chantilly Nov 30 02:44:54 DEBUG[4684]: app_disa.c:165 disa_exec: DISA no-password login success Nov 30 02:44:54 DEBUG[4684]: channel.c:1128 ast_settimeout: Scheduling timer at 160 sample intervals Nov 30 02:44:54 DEBUG[4684]: channel.c:1379 ast_read: Generator got voice, switching to phase locked mode Nov 30 02:44:54 DEBUG[4684]: channel.c:1128 ast_settimeout: Scheduling timer at 0 sample intervals Nov 30 02:44:54 DEBUG[4684]: chan_zap.c:4031 zt_read...

...class default requested but no musiconhold loaded. [Oct 2 11:09:21] DEBUG[29533]: channel.c:3616 set_format: Set channel SIP/1000-0a292360 to write format gsm [Oct 2 11:09:21] DEBUG[29533]: rtp.c:3099 ast_rtp_raw_write: Difference is 1952, ms is 264 [Oct 2 11:09:21] DEBUG[29533]: channel.c:2362 ast_settimeout: Scheduling timer at 160 sample intervals -- <SIP/1000-0a292360> Playing 'pbx-transfer.gsm' (language 'en') [Oct 2 11:09:21] DEBUG[29533]: rtp.c:958 process_rfc2833: - RTP 2833 Event: 00000008 (len = 4) [Oct 2 11:09:21] DEBUG[29533]: rtp.c:958 process_rfc2833: - RTP 2833...

...class default requested but no musiconhold loaded. [Oct 2 11:09:21] DEBUG[29533]: channel.c:3616 set_format: Set channel SIP/1000-0a292360 to write format gsm [Oct 2 11:09:21] DEBUG[29533]: rtp.c:3099 ast_rtp_raw_write: Difference is 1952, ms is 264 [Oct 2 11:09:21] DEBUG[29533]: channel.c:2362 ast_settimeout: Scheduling timer at 160 sample intervals -- <SIP/1000-0a292360> Playing 'pbx-transfer.gsm' (language 'en') [Oct 2 11:09:21] DEBUG[29533]: rtp.c:958 process_rfc2833: - RTP 2833 Event: 00000008 (len = 4) [Oct 2 11:09:21] DEBUG[29533]: rtp.c:958 process_rfc2833: - RTP 2833...

...class default requested but no musiconhold loaded. [Oct 2 11:09:21] DEBUG[29533]: channel.c:3616 set_format: Set channel SIP/1000-0a292360 to write format gsm [Oct 2 11:09:21] DEBUG[29533]: rtp.c:3099 ast_rtp_raw_write: Difference is 1952, ms is 264 [Oct 2 11:09:21] DEBUG[29533]: channel.c:2362 ast_settimeout: Scheduling timer at 160 sample intervals -- <SIP/1000-0a292360> Playing 'pbx-transfer.gsm' (language 'en') [Oct 2 11:09:21] DEBUG[29533]: rtp.c:958 process_rfc2833: - RTP 2833 Event: 00000008 (len = 4) [Oct 2 11:09:21] DEBUG[29533]: rtp.c:958 process_rfc2833: - RTP 2833...

...class default requested but no musiconhold loaded. [Oct 2 11:09:21] DEBUG[29533]: channel.c:3616 set_format: Set channel SIP/1000-0a292360 to write format gsm [Oct 2 11:09:21] DEBUG[29533]: rtp.c:3099 ast_rtp_raw_write: Difference is 1952, ms is 264 [Oct 2 11:09:21] DEBUG[29533]: channel.c:2362 ast_settimeout: Scheduling timer at 160 sample intervals -- <SIP/1000-0a292360> Playing 'pbx-transfer.gsm' (language 'en') [Oct 2 11:09:21] DEBUG[29533]: rtp.c:958 process_rfc2833: - RTP 2833 Event: 00000008 (len = 4) [Oct 2 11:09:21] DEBUG[29533]: rtp.c:958 process_rfc2833: - RTP 2833...

...261 pbx_extension_helper: Launching 'BackGround' -- Executing BackGround("Zap/1-1", "Sysnux/bonjour1") in new stack Dec 17 11:22:32 DEBUG[3600]: channel.c:1710 ast_set_write_format: Set channel Zap/1-1 to write format slin Dec 17 11:22:32 DEBUG[3600]: channel.c:1131 ast_settimeout: Scheduling timer at 160 sample intervals -- Playing 'Sysnux/bonjour1' (language 'fr') Urgent handler Dec 17 11:22:40 DEBUG[3600]: channel.c:1131 ast_settimeout: Scheduling timer at 97 sample intervals Dec 17 11:22:40 DEBUG[3600]: channel.c:1131 ast_settimeout: Scheduling timer...

...-- Set Digit Timeout to 5 -- Executing ResponseTimeout("Modem[i4l]/ttyI0", "10") in new stack -- Set Response Timeout to 10 -- Executing BackGround("Modem[i4l]/ttyI0", "demo-congrats") in new stack Sep 23 01:12:07 DEBUG[1111542704]: channel.c:1128 ast_settimeout: Scheduling timer at 160 sample intervals -- Playing 'demo-congrats' (language 'en') Sep 23 01:12:08 DEBUG[1111542704]: chan_modem_i4l.c:413 i4l_read: Value of escape is ^ (3)... Sep 23 01:12:08 DEBUG[1111542704]: channel.c:1128 ast_settimeout: Scheduling timer at 0 sample inter...

...y that doesn't help me and I want to see why one system works and one doesn't This is dtmf debug from an iax handset sending digit 4 [Jan 12 23:13:55] DEBUG[8717]: channel.c:3372 set_format: Set channel SIP/xtreme-00000639 to write format slin [Jan 12 23:13:55] DEBUG[8717]: channel.c:1986 ast_settimeout: Scheduling timer at 160 sample intervals [Jan 12 23:13:55] DEBUG[8717]: channel.c:5297 ast_channel_start_silence_generator: Started silence generator on 'SIP/xtreme-00000639' [Jan 12 23:13:55] DEBUG[8717]: rtp.c:2796 ast_rtp_raw_write: Difference is 1736, ms is 237 [Jan 12 23:13:55] DEBUG[...

...two keypresses (seems like two, with the "Oooh got something" message being asterisk's way of saying at least on extension exists in the context that begins with a 1). DEBUG[14351]: File chan_zap.c, Line 1872 (zt_answer): Took Zap/27-1 off hook DEBUG[14351]: File channel.c, Line 952 (ast_settimeout): Scheduling timer at 160 sample intervals DEBUG[14351]: File chan_zap.c, Line 3388 (zt_read): DTMF digit: 1 on Zap/27-1 DEBUG[14351]: File channel.c, Line 952 (ast_settimeout): Scheduling timer at 0 sample intervals DEBUG[14351]: File pbx.c, Line 1686 (ast_pbx_run): Oooh, got something to jump out...

...channel 'SIP/192.168.1.38-082257a0' Jan 25 17:34:04 DEBUG[10020]: channel.c:1707 ast_set_write_format: Set channel SIP/192.168.1.38-082257a0 to write format slin -- Started music on hold, class 'default', on SIP/192.168.1.38-082257a0 Jan 25 17:34:04 DEBUG[10020]: channel.c:1128 ast_settimeout: Scheduling timer at 160 sample intervals Urgent handler Jan 25 17:34:04 DEBUG[10020]: channel.c:1379 ast_read: Generator got voice, switching to phase locked mode Jan 25 17:34:04 DEBUG[10020]: channel.c:1128 ast_settimeout: Scheduling timer at 0 sample intervals Jan 25 17:34:04 DEBUG[10020]: rtp.c...

...s ringing Aug 2 11:04:22 DEBUG[557083]: channel.c:1436 ast_indicate: Driver for channel 'SIP/4001-40ee' does not support indication 3, emulating it Aug 2 11:04:22 DEBUG[557083]: channel.c:1551 ast_prod: Prodding channel 'SIP/4001-40ee' Aug 2 11:04:22 DEBUG[557083]: channel.c:1128 ast_settimeout: Scheduling timer at 160 sample intervals Aug 2 11:04:22 DEBUG[557083]: channel.c:1128 ast_settimeout: Scheduling timer at 0 sample intervals -- Channel 0/1, span 1 got hangup Aug 2 11:04:28 DEBUG[557083]: chan_zap.c:2427 zt_setoption: Set option AUDIO MODE, value: ON(1) on Zap/1-1 Aug 2 11:...

...n new stack -- Set Response Timeout to 30 -- Executing BackGround("SIP/s00227156-a5ef", "welcome-mainmenu") in new stack Aug 24 23:14:33 DEBUG[-1221325904]: rtp.c:1146 ast_rtp_write: Ooh, format changed from UNKN to ULAW Aug 24 23:14:33 DEBUG[-1221325904]: channel.c:1101 ast_settimeout: Scheduling timer at 160 sample intervals -- Playing 'welcome-mainmenu' (language 'en') Aug 24 23:14:33 DEBUG[-1126876240]: chan_sip.c:817 __sip_ack: Stopping retransmission on '5bd8e9e869d4a8306d256fa02f387e09@66.234.228.137' of Response 103: Found Aug 24 23:14:37 DEBUG...

Hi, I am developing asterisk support for our application using the Async AGI and Asterisk-Java. One thing I haven't been able to implement is how to stop playing a sound. Something similar to StopIO for Dialogic GlobalCall or DivaStopSending for Eicon. Is there any way to achieve this today which I have missed? Or could someone give me hints on how I could implement this in the res_agi.c The

...Checking SI P call limits for device Phone3 Sep 13 22:18:10 DEBUG[13167]: chan_sip.c:5497 build_route: build_route: Contact hop: <sip:Phone3@192.168.0.18:5061> -- Executing VoiceMailMain("SIP/Phone3-9d74", "") in new stack Sep 13 22:18:10 DEBUG[13167]: channel.c:1388 ast_settimeout: Scheduling timer at 160 sample intervals -- Playing 'vm-login' (language 'en') Sep 13 22:18:10 DEBUG[13167]: chan_sip.c:1274 __sip_ack: Stopping retransmission on '8CB86803-55AC-4263-BB6F-01E0E056EC1B@192.168.0.18' of Response 30127: Match Found Sep 13 22:18:1...

...pbx_extension_helper: Launching 'Playback' -- Executing Playback("Zap/1-1", "demo-congrats") in new stack Urgent handler Aug 4 15:43:40 DEBUG[5059]: channel.c:1719 ast_set_write_format: Set channel Zap/1-1 to write format gsm Aug 4 15:43:40 DEBUG[5059]: channel.c:1128 ast_settimeout: Scheduling timer at 160 sample intervals -- Playing 'demo-congrats' (language 'en') Urgent handler Aug 4 15:44:08 DEBUG[5059]: channel.c:1128 ast_settimeout: Scheduling timer at 0 sample intervals Aug 4 15:44:08 DEBUG[5059]: channel.c:1128 ast_settimeout: Scheduling timer at 0 s...