Re: [asterisk-users] Problem with extensions in IVR and queues
Hi, we've just been able to find the problem. Apparently it was related to the softphone. We've installed another one and the call is performed ok. Thanks! Anahi Ludueña From: a_ludu...@hotmail.com To: asterisk-users@lists.digium.com Date: Wed, 30 Jun 2010 19:59:14 + Subject: Re: [asterisk-users] Problem with extensions in IVR and queues Ups, sorry, that CLI output is related to my other problem (the options of IVR doesn't responde when the call is from landline or cell phone). I'll put the correct CLI output... Thanks, Anahi Ludueña From: a_ludu...@hotmail.com To: asterisk-users@lists.digium.com Date: Wed, 30 Jun 2010 19:50:00 + Subject: Re: [asterisk-users] Problem with extensions in IVR and queues This is the CLI output, the dialplan is the one that the Elastix creates when somebody sets the followme... I don't know what part you want I post here... Thanks, -- Executing [4...@from-internal:1] GotoIf(SIP/9050-001185aa, 0?ext-local|4010|1) in new stack -- Executing [4...@from-internal:2] Macro(SIP/9050-001185aa, user-callerid|) in new stack -- Executing [...@macro-user-callerid:1] Set(SIP/9050-001185aa, AMPUSER=9050) in new stack -- Executing [...@macro-user-callerid:2] GotoIf(SIP/9050-001185aa, 0?report) in new stack -- Executing [...@macro-user-callerid:3] ExecIf(SIP/9050-001185aa, 1|Set|REALCALLERIDNUM=9050) in new stack -- Executing [...@macro-user-callerid:4] Set(SIP/9050-001185aa, AMPUSER=9050) in new stack -- Executing [...@macro-user-callerid:5] Set(SIP/9050-001185aa, AMPUSERCIDNAME=CALLPBX) in new stack -- Executing [...@macro-user-callerid:6] GotoIf(SIP/9050-001185aa, 0?report) in new stack -- Executing [...@macro-user-callerid:7] Set(SIP/9050-001185aa, AMPUSERCID=9050) in new stack -- Executing [...@macro-user-callerid:8] Set(SIP/9050-001185aa, CALLERID(all)=CALLPBX 9050) in new stack -- Executing [...@macro-user-callerid:9] ExecIf(SIP/9050-001185aa, 0|Set|CHANNEL(language)=) in new stack -- Executing [...@macro-user-callerid:10] GotoIf(SIP/9050-001185aa, 0?continue) in new stack -- Executing [...@macro-user-callerid:11] Set(SIP/9050-001185aa, __TTL=64) in new stack -- Executing [...@macro-user-callerid:12] GotoIf(SIP/9050-001185aa, 1?continue) in new stack -- Goto (macro-user-callerid,s,19) -- Executing [...@macro-user-callerid:19] NoOp(SIP/9050-001185aa, Using CallerID CALLPBX 9050) in new stack -- Executing [4...@from-internal:3] GotoIf(SIP/9050-001185aa, 1?skipdb) in new stack -- Goto (from-internal,4010,5) -- Executing [4...@from-internal:5] Set(SIP/9050-001185aa, __NODEST=) in new stack -- Executing [4...@from-internal:6] Set(SIP/9050-001185aa, __BLKVM_OVERRIDE=BLKVM/4010/SIP/9050-001185aa) in new stack -- Executing [4...@from-internal:7] Set(SIP/9050-001185aa, __BLKVM_BASE=4010) in new stack -- Executing [4...@from-internal:8] Set(SIP/9050-001185aa, DB(BLKVM/4010/SIP/9050-001185aa)=TRUE) in new stack -- Executing [4...@from-internal:9] Set(SIP/9050-001185aa, RRNODEST=) in new stack -- Executing [4...@from-internal:10] Set(SIP/9050-001185aa, __NODEST=4010) in new stack -- Executing [4...@from-internal:11] Set(SIP/9050-001185aa, RecordMethod=Group) in new stack -- Executing [4...@from-internal:12] Macro(SIP/9050-001185aa, record-enable|4010|Group) in new stack -- Executing [...@macro-record-enable:1] GotoIf(SIP/9050-001185aa, 1?check) in new stack -- Goto (macro-record-enable,s,4) -- Executing [...@macro-record-enable:4] AGI(SIP/9050-001185aa, recordingcheck|20100630-154030|1277926830.37214) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck -- AGI Script recordingcheck completed, returning 0 -- Executing [...@macro-record-enable:5] MacroExit(SIP/9050-001185aa, ) in new stack -- Executing [4...@from-internal:13] Set(SIP/9050-001185aa, RingGroupMethod=ringallv2) in new stack -- Executing [4...@from-internal:14] Set(SIP/9050-001185aa, _FMGRP=4010) in new stack -- Executing [4...@from-internal:15] GotoIf(SIP/9050-001185aa, 0?doconfirm) in new stack -- Executing [4...@from-internal:16] Macro(SIP/9050-001185aa, dial|20|tr|4010) in new stack -- Executing [...@macro-dial:1] GotoIf(SIP/9050-001185aa, 1?dial) in new stack -- Goto (macro-dial,s,3) -- Executing [...@macro-dial:3] AGI(SIP/9050-001185aa, dialparties.agi) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi dialparties.agi: Starting New Dialparties.agi == Parsing '/etc/asterisk/manager.conf': Found == Parsing '/etc/asterisk/manager_additional.conf': Found == Parsing '/etc/asterisk/manager_custom.conf': Found == Manager 'admin' logged on from 127.0.0.1 dialparties.agi: Caller ID name is 'CALLPBX' number is '9050' dialparties.agi: USE_CONFIRMATION: 'FALSE' dialparties.agi: RINGGROUP_INDEX: '' dialparties.agi: Methodology of ring
Re: [asterisk-users] Dial options not working
Hi, do you mean what kind of extension I have? it is SIP, but from it, everything works well... In the SIP extension, the DTMF mode is rfc2833. Thanks, From: asteriskus...@dovid.net To: asterisk-users@lists.digium.com Date: Wed, 30 Jun 2010 13:54:50 +0300 Subject: Re: [asterisk-users] Dial options not working Anahi, What kind of line do you have ? POTS, PRI, SIP ? It seems like the DTMF is not coming in correctly or you have some bad settings on your end. - Original Message - From: Anahi Ludueña To: asterisk-users@lists.digium.com Sent: Wednesday, June 30, 2010 01:17 Subject: Re: [asterisk-users] Dial options not working Thanks, but I don't have any *dahdi*.conf file here... (I check in /etc/asterisk) Anahi Ludueña From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Tue, 29 Jun 2010 16:54:01 -0500 Subject: Re: [asterisk-users] Dial options not working Check your DTMF settings in *dahdi*.conf (not sure which of the dahdi files this lives in). Sounds like your DAHDI doesn’t like DTMF input. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Anahi Ludueña Sent: Tuesday, June 29, 2010 4:51 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Dial options not working Hi, I have an extension which has the follow me option activated. The followme option should go to a IVR if no answer... The problem that I have is that everything works when I'm calling it from my extension, but if I use any landline phone or a cell phone, I'm unable to enter any options. When I press one option, it seems I do nothing... Please, could you help me? Thanks, Anahi Ludueña Disfruta de Hotmail y Messenger en tu móvil con YOIGO. ¡Hazlo ya! Dime cómo viajas y te diré qué famoso eres ¿Cuál es tu estilo, chic y deslumbrante o mundano y familiar? Descubre quién eres viajando. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ ¿Quieres descubrir todos los trucos de Windows 7? ¡Hazlo aquí! http://www.sietesunpueblodeexpertos.com/index_windows7.html-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dial options not working
Hi, yes, I've just tried to use the dtmf mode inband, but it doesn't work with landline phones or cell phones... Thanks, Anahi Ludueña Date: Wed, 30 Jun 2010 12:56:59 +0100 From: kwat...@geniusgroupltd.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Dial options not working Hi, Have you tried sending the dtmf inband? I've had more success interoping betwen different vendors with inband DTMF. Thanks Kenny Watson From: Anahi Ludueña a_ludu...@hotmail.com To: asterisk-users@lists.digium.com Sent: Wednesday, 30 June, 2010 12:50:23 PM Subject: Re: [asterisk-users] Dial options not working Hi, do you mean what kind of extension I have? it is SIP, but from it, everything works well... In the SIP extension, the DTMF mode is rfc2833. Thanks, From: asteriskus...@dovid.net To: asterisk-users@lists.digium.com Date: Wed, 30 Jun 2010 13:54:50 +0300 Subject: Re: [asterisk-users] Dial options not working Anahi, What kind of line do you have ? POTS, PRI, SIP ? It seems like the DTMF is not coming in correctly or you have some bad settings on your end. - Original Message - From: Anahi Ludueña To: asterisk-users@lists.digium.com Sent: Wednesday, June 30, 2010 01:17 Subject: Re: [asterisk-users] Dial options not working Thanks, but I don't have any *dahdi*.conf file here... (I check in /etc/asterisk) Anahi Ludueña From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Tue, 29 Jun 2010 16:54:01 -0500 Subject: Re: [asterisk-users] Dial options not working Check your DTMF settings in *dahdi*.conf (not sure which of the dahdi files this lives in). Sounds like your DAHDI doesn’t like DTMF input. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Anahi Ludueña Sent: Tuesday, June 29, 2010 4:51 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Dial options not working Hi, I have an extension which has the follow me option activated. The followme option should go to a IVR if no answer... The problem that I have is that everything works when I'm calling it from my extension, but if I use any landline phone or a cell phone, I'm unable to enter any options. When I press one option, it seems I do nothing... Please, could you help me? Thanks, Anahi Ludueña Disfruta de Hotmail y Messenger en tu móvil con YOIGO. ¡Hazlo ya! Dime cómo viajas y te diré qué famoso eres ¿Cuál es tu estilo, chic y deslumbrante o mundano y familiar? Descubre quién eres viajando. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Del Lado Oscuro de Internet protegerte puedes. ¡Entra ya en www.ayudartepodria.com! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Sé el protagonista de GQ con Messenger y Vodafone Blackberry. ¡Y gana premios! http://serviciosmoviles.es.msn.com/messenger/vodafone.aspx-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem with extensions in IVR and queues
Hi people, we have some extensions which are included in the IVRs and/or queues. Everything works fine, but the calls done from these extensions are hang up after 30 o 35 seconds. If they are not included in the IVR or queues, the calls are performed well. Do you know if there is something else to set? Thanks, Anahi Ludueña _ ¿Un navegador seguro buscando estás? ¡Protegete ya en www.ayudartepodria.com! www.ayudartepodria.com-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with extensions in IVR and queues
Thanks Danny, but I don't know what I should do to fix it... Could you help me? Anahi Ludueña From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Wed, 30 Jun 2010 10:33:31 -0500 Subject: Re: [asterisk-users] Problem with extensions in IVR and queues Sounds like you are getting a “dial without bridge” – asterisk dials x and make the connection, but because the bridge doesn’t happen for what ever reason, the call disconnects like no one ever answered. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Anahi Ludueña Sent: Wednesday, June 30, 2010 10:29 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Problem with extensions in IVR and queues Hi people, we have some extensions which are included in the IVRs and/or queues. Everything works fine, but the calls done from these extensions are hang up after 30 o 35 seconds. If they are not included in the IVR or queues, the calls are performed well. Do you know if there is something else to set? Thanks, Anahi Ludueña ¿Un navegador seguro buscando estás? ¡Protegete ya en www.ayudartepodria.com! _ Citas sin compromiso por Internet Te damos las claves para encontrar pareja en la red http://contactos.es.msn.com/?mtcmk=015352-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with extensions in IVR and queues
-- dialparties.agi: Filtered ARG3: 4010 dialparties.agi: NODEST: 4010 adding M(auto-blkvm) to dialopts: trM(auto-blkvm) dialparties.agi: NODEST: 4010 blkvm enabled macro already in dialopts: trM(auto-blkvm) == Manager 'admin' logged off from 127.0.0.1 -- AGI Script dialparties.agi completed, returning 0 -- Executing [...@macro-dial:7] Dial(SIP/9050-001185aa, SIP/4010|22|trM(auto-blkvm)) in new stack Really destroying SIP dialog '1544c4ea374acd44596154e42c848...@127.0.0.1' Method: INVITE == Everyone is busy/congested at this time (1:0/0/1) -- Executing [...@macro-dial:8] Set(SIP/9050-001185aa, DIALSTATUS=CHANUNAVAIL) in new stack -- Executing [...@macro-dial:9] GosubIf(SIP/9050-001185aa, 0?CHANUNAVAIL|1) in new stack -- Executing [4...@from-internal:17] Goto(SIP/9050-001185aa, nextstep) in new stack -- Goto (from-internal,4010,19) -- Executing [4...@from-internal:19] Set(SIP/9050-001185aa, RingGroupMethod=) in new stack -- Executing [4...@from-internal:20] GotoIf(SIP/9050-001185aa, 0?nodest) in new stack -- Executing [4...@from-internal:21] Set(SIP/9050-001185aa, __NODEST=) in new stack -- Executing [4...@from-internal:22] DBdel(SIP/9050-001185aa, BLKVM/4010/SIP/9050-001185aa) in new stack -- DBdel: family=BLKVM, key=4010/SIP/9050-001185aa -- Executing [4...@from-internal:23] Goto(SIP/9050-001185aa, ivr-3|s|1) in new stack -- Goto (ivr-3,s,1) -- Executing [...@ivr-3:1] Set(SIP/9050-001185aa, MSG=custom/CALL-English) in new stack -- Executing [...@ivr-3:2] Set(SIP/9050-001185aa, LOOPCOUNT=0) in new stack -- Executing [...@ivr-3:3] Set(SIP/9050-001185aa, __DIR-CONTEXT=default) in new stack -- Executing [...@ivr-3:4] Set(SIP/9050-001185aa, _IVR_CONTEXT_ivr-3=) in new stack -- Executing [...@ivr-3:5] Set(SIP/9050-001185aa, _IVR_CONTEXT=ivr-3) in new stack -- Executing [...@ivr-3:6] GotoIf(SIP/9050-001185aa, 0?begin) in new stack -- Executing [...@ivr-3:7] Answer(SIP/9050-001185aa, ) in new stack -- Executing [...@ivr-3:8] Wait(SIP/9050-001185aa, 1) in new stack -- Executing [...@ivr-3:9] Set(SIP/9050-001185aa, TIMEOUT(digit)=3) in new stack -- Digit timeout set to 3 -- Executing [...@ivr-3:10] Set(SIP/9050-001185aa, TIMEOUT(response)=10) in new stack -- Response timeout set to 10 -- Executing [...@ivr-3:11] Set(SIP/9050-001185aa, __IVR_RETVM=) in new stack -- Executing [...@ivr-3:12] ExecIf(SIP/9050-001185aa, 1|Background|custom/CALL-English) in new stack -- SIP/9050-001185aa Playing 'custom/CALL-English' (language 'en') Really destroying SIP dialog '48d34342645adfa70265fa8e5291c...@xxx.xxx.xxx.xxx' Method: OPTIONS Really destroying SIP dialog '200bf37a463ff4bb5673ba4720cec...@xxx.xxx.xxx.xxx' Method: OPTIONS Really destroying SIP dialog '24d9c31a44a206f216d2c142338fb...@xxx.xxx.xxx.xxx' Method: NOTIFY -- Got SIP response 603 Declined (no dialog) back from YYY.YYY.YYY.YYY Really destroying SIP dialog '42debf4b37838b98708590dc6e425...@xxx.xxx.xxx.xxx' Method: NOTIFY -- Executing [...@ivr-3:13] WaitExten(SIP/9050-001185aa, |) in new stack == Spawn extension (ivr-3, s, 13) exited non-zero on 'SIP/9050-001185aa' -- Executing [...@ivr-3:1] Hangup(SIP/9050-001185aa, ) in new stack == Spawn extension (ivr-3, h, 1) exited non-zero on 'SIP/9050-001185aa' Anahi Ludueña From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Wed, 30 Jun 2010 14:08:19 -0500 Subject: Re: [asterisk-users] Problem with extensions in IVR and queues Can you post the dialplan section and CLI output from one of these calls? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Anahi Ludueña Sent: Wednesday, June 30, 2010 2:05 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Problem with extensions in IVR and queues Thanks Danny, but I don't know what I should do to fix it... Could you help me? Anahi Ludueña From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Wed, 30 Jun 2010 10:33:31 -0500 Subject: Re: [asterisk-users] Problem with extensions in IVR and queues Sounds like you are getting a “dial without bridge” – asterisk dials x and make the connection, but because the bridge doesn’t happen for what ever reason, the call disconnects like no one ever answered. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Anahi Ludueña Sent: Wednesday, June 30, 2010 10:29 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Problem with extensions in IVR and queues Hi people, we have some extensions which are included in the IVRs and/or queues. Everything works fine, but the calls done from these extensions are hang up after 30 o 35 seconds. If they are not included in the IVR or queues, the calls
Re: [asterisk-users] Problem with extensions in IVR and queues
Ups, sorry, that CLI output is related to my other problem (the options of IVR doesn't responde when the call is from landline or cell phone). I'll put the correct CLI output... Thanks, Anahi Ludueña From: a_ludu...@hotmail.com To: asterisk-users@lists.digium.com Date: Wed, 30 Jun 2010 19:50:00 + Subject: Re: [asterisk-users] Problem with extensions in IVR and queues This is the CLI output, the dialplan is the one that the Elastix creates when somebody sets the followme... I don't know what part you want I post here... Thanks, -- Executing [4...@from-internal:1] GotoIf(SIP/9050-001185aa, 0?ext-local|4010|1) in new stack -- Executing [4...@from-internal:2] Macro(SIP/9050-001185aa, user-callerid|) in new stack -- Executing [...@macro-user-callerid:1] Set(SIP/9050-001185aa, AMPUSER=9050) in new stack -- Executing [...@macro-user-callerid:2] GotoIf(SIP/9050-001185aa, 0?report) in new stack -- Executing [...@macro-user-callerid:3] ExecIf(SIP/9050-001185aa, 1|Set|REALCALLERIDNUM=9050) in new stack -- Executing [...@macro-user-callerid:4] Set(SIP/9050-001185aa, AMPUSER=9050) in new stack -- Executing [...@macro-user-callerid:5] Set(SIP/9050-001185aa, AMPUSERCIDNAME=CALLPBX) in new stack -- Executing [...@macro-user-callerid:6] GotoIf(SIP/9050-001185aa, 0?report) in new stack -- Executing [...@macro-user-callerid:7] Set(SIP/9050-001185aa, AMPUSERCID=9050) in new stack -- Executing [...@macro-user-callerid:8] Set(SIP/9050-001185aa, CALLERID(all)=CALLPBX 9050) in new stack -- Executing [...@macro-user-callerid:9] ExecIf(SIP/9050-001185aa, 0|Set|CHANNEL(language)=) in new stack -- Executing [...@macro-user-callerid:10] GotoIf(SIP/9050-001185aa, 0?continue) in new stack -- Executing [...@macro-user-callerid:11] Set(SIP/9050-001185aa, __TTL=64) in new stack -- Executing [...@macro-user-callerid:12] GotoIf(SIP/9050-001185aa, 1?continue) in new stack -- Goto (macro-user-callerid,s,19) -- Executing [...@macro-user-callerid:19] NoOp(SIP/9050-001185aa, Using CallerID CALLPBX 9050) in new stack -- Executing [4...@from-internal:3] GotoIf(SIP/9050-001185aa, 1?skipdb) in new stack -- Goto (from-internal,4010,5) -- Executing [4...@from-internal:5] Set(SIP/9050-001185aa, __NODEST=) in new stack -- Executing [4...@from-internal:6] Set(SIP/9050-001185aa, __BLKVM_OVERRIDE=BLKVM/4010/SIP/9050-001185aa) in new stack -- Executing [4...@from-internal:7] Set(SIP/9050-001185aa, __BLKVM_BASE=4010) in new stack -- Executing [4...@from-internal:8] Set(SIP/9050-001185aa, DB(BLKVM/4010/SIP/9050-001185aa)=TRUE) in new stack -- Executing [4...@from-internal:9] Set(SIP/9050-001185aa, RRNODEST=) in new stack -- Executing [4...@from-internal:10] Set(SIP/9050-001185aa, __NODEST=4010) in new stack -- Executing [4...@from-internal:11] Set(SIP/9050-001185aa, RecordMethod=Group) in new stack -- Executing [4...@from-internal:12] Macro(SIP/9050-001185aa, record-enable|4010|Group) in new stack -- Executing [...@macro-record-enable:1] GotoIf(SIP/9050-001185aa, 1?check) in new stack -- Goto (macro-record-enable,s,4) -- Executing [...@macro-record-enable:4] AGI(SIP/9050-001185aa, recordingcheck|20100630-154030|1277926830.37214) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck -- AGI Script recordingcheck completed, returning 0 -- Executing [...@macro-record-enable:5] MacroExit(SIP/9050-001185aa, ) in new stack -- Executing [4...@from-internal:13] Set(SIP/9050-001185aa, RingGroupMethod=ringallv2) in new stack -- Executing [4...@from-internal:14] Set(SIP/9050-001185aa, _FMGRP=4010) in new stack -- Executing [4...@from-internal:15] GotoIf(SIP/9050-001185aa, 0?doconfirm) in new stack -- Executing [4...@from-internal:16] Macro(SIP/9050-001185aa, dial|20|tr|4010) in new stack -- Executing [...@macro-dial:1] GotoIf(SIP/9050-001185aa, 1?dial) in new stack -- Goto (macro-dial,s,3) -- Executing [...@macro-dial:3] AGI(SIP/9050-001185aa, dialparties.agi) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi dialparties.agi: Starting New Dialparties.agi == Parsing '/etc/asterisk/manager.conf': Found == Parsing '/etc/asterisk/manager_additional.conf': Found == Parsing '/etc/asterisk/manager_custom.conf': Found == Manager 'admin' logged on from 127.0.0.1 dialparties.agi: Caller ID name is 'CALLPBX' number is '9050' dialparties.agi: USE_CONFIRMATION: 'FALSE' dialparties.agi: RINGGROUP_INDEX: '' dialparties.agi: Methodology of ring is 'ringallv2' -- dialparties.agi: Added extension 4010 to extension map dialparties.agi: got fmgrp_prering: 2, fmgrp_grptime: 20 dialparties.agi: fmgrp_totalprering: 22 dialparties.agi: found extension in pre-ring and array dialparties.agi: ringallv2 ring times: REALPRERING: 22, PRERING: 2 -- dialparties.agi
[asterisk-users] Dial options not working
Hi, I have an extension which has the follow me option activated. The followme option should go to a IVR if no answer... The problem that I have is that everything works when I'm calling it from my extension, but if I use any landline phone or a cell phone, I'm unable to enter any options. When I press one option, it seems I do nothing... Please, could you help me? Thanks, Anahi Ludueña _ Sé el protagonista de GQ con Messenger y Vodafone Blackberry. ¡Y gana premios! http://serviciosmoviles.es.msn.com/messenger/vodafone.aspx-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dial options not working
Thanks, but I don't have any *dahdi*.conf file here... (I check in /etc/asterisk) Anahi Ludueña From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Tue, 29 Jun 2010 16:54:01 -0500 Subject: Re: [asterisk-users] Dial options not working Check your DTMF settings in *dahdi*.conf (not sure which of the dahdi files this lives in). Sounds like your DAHDI doesn’t like DTMF input. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Anahi Ludueña Sent: Tuesday, June 29, 2010 4:51 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Dial options not working Hi, I have an extension which has the follow me option activated. The followme option should go to a IVR if no answer... The problem that I have is that everything works when I'm calling it from my extension, but if I use any landline phone or a cell phone, I'm unable to enter any options. When I press one option, it seems I do nothing... Please, could you help me? Thanks, Anahi Ludueña Disfruta de Hotmail y Messenger en tu móvil con YOIGO. ¡Hazlo ya! _ Citas sin compromiso por Internet Te damos las claves para encontrar pareja en la red http://contactos.es.msn.com/?mtcmk=015352-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] FW: Music on Hold problema
Please, I need help with this... Anahi Ludueña From: a_ludu...@hotmail.com To: asterisk-users@lists.digium.com Date: Fri, 18 Jun 2010 15:12:25 + Subject: Re: [asterisk-users] Music on Hold problema The list of /var/lib/asterisk/mohmp3 is: -rw-rw 4 asterisk asterisk 184 Oct 19 2009 LICENSE-asterisk-moh-freeplay-wav -rw-rw-r-- 4 asterisk asterisk 882748 Oct 19 2009 QuajiroPromo.sln -rw-rw-r-- 4 asterisk asterisk 834682 Oct 19 2009 TristeAlegriaPromo.sln -rw-rw 4 asterisk asterisk 1939794 Oct 19 2009 fpm-calm-river.wav -rw-rw 4 asterisk asterisk 2582196 Oct 19 2009 fpm-sunshine.wav -rw-rw 4 asterisk asterisk 2217318 Oct 19 2009 fpm-world-mix.wav And the musiconhold.conf is: [default] mode=files directory=/var/lib/asterisk/mohmp3 random=yes [none] mode=files directory=/dev/null Thanks, Anahi Ludueña From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Fri, 18 Jun 2010 09:26:16 -0500 Subject: Re: [asterisk-users] Music on Hold problema Post the /var/lib/asterisk/mohmp3 listing and musiconhold.conf From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Anahi Ludueña Sent: Friday, June 18, 2010 9:18 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Music on Hold problema Any ideas, please? Anahi Ludueña From: a_ludu...@hotmail.com To: asterisk-users@lists.digium.com Date: Thu, 17 Jun 2010 19:54:30 + Subject: Re: [asterisk-users] Music on Hold problema I have wav files in the /var/lib/asterisk/mohmp3... Anahi Ludueña From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Thu, 17 Jun 2010 14:36:00 -0500 Subject: Re: [asterisk-users] Music on Hold problema I see that moh is trying sln format, then ulaw, then failing. Do you have moh files in either of these formats? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Anahi Ludueña Sent: Thursday, June 17, 2010 2:24 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Music on Hold problema Hi people, I have a problem with Music On Hold, it is stopped just after starting... This is the log: [Jun 17 19:20:22] VERBOSE[20784] logger.c: -- Executing [...@test:1] NoOp(SIP/7PBX-08229d18, Start) in new stack [Jun 17 19:20:22] DEBUG[20784] pbx.c: Launching 'MusicOnHold' [Jun 17 19:20:22] VERBOSE[20784] logger.c: -- Executing [...@test:2] MusicOnHold(SIP/7PBX-08229d18, ) in new stack [Jun 17 19:20:22] DEBUG[20784] channel.c: Set channel SIP/7PBX-08229d18 to write format slin [Jun 17 19:20:22] VERBOSE[20784] logger.c: -- Started music on hold, class 'default', on channel 'SIP/7PBX-08229d18' [Jun 17 19:20:22] DEBUG[20784] channel.c: Scheduling timer at 160 sample intervals [Jun 17 19:20:22] DEBUG[20784] channel.c: Auto-deactivating generator [Jun 17 19:20:22] DEBUG[20784] channel.c: Set channel SIP/7PBX-08229d18 to write format ulaw [Jun 17 19:20:22] VERBOSE[20784] logger.c: -- Stopped music on hold on SIP/7PBX-08229d18 [Jun 17 19:20:22] DEBUG[20784] channel.c: Scheduling timer at 0 sample intervals Could you help me with this? Thanks, Anahi Ludueña Disfruta de Hotmail y Messenger en tu móvil con YOIGO. ¡Hazlo ya! ¿Un navegador seguro buscando estás? ¡Protegete ya en www.ayudartepodria.com! Del Lado Oscuro de Internet protegerte puedes. ¡Entra ya en www.ayudartepodria.com! Del Lado Oscuro de Internet protegerte puedes. ¡Entra ya en www.ayudartepodria.com! _ Sé el protagonista de GQ con Messenger y Vodafone Blackberry. ¡Y gana premios! http://serviciosmoviles.es.msn.com/messenger/vodafone.aspx-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FW: Music on Hold problema
One thing to take into account and I haven't said before, sorry... I have 2 pbx, one is connecting to the other by a SIP trunk... The first pbx has the setting which I put some days ago... the second pbx has the extensions and I'm trying to use them in the call. Everything is working, except the music on hold. Thanks, Anahi Ludueña Date: Wed, 23 Jun 2010 10:44:10 -0400 From: zisha...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] FW: Music on Hold problema The moh conf file seems good. It is the standard implementation and should have worked. Just wondering if your end devices, whether they are IP phones or softphones, are setup to listen to some different codecs than ulaw and slin? Or in your sip.conf when declaring extensions you are not putting the correct codecs in the 'allow=' declaration. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-06-23 10:33 AM, Anahi Ludueña a_ludu...@hotmail.com wrote: Please, I need help with this... Anahi Ludueña From: a_... Date: Fri, 18 Jun 2010 15:12:25 + Subject: Re: [asterisk-users] Music on Hold problema The list of /var/lib/asterisk/mohmp3 is: -rw... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Los cochazos de los famosos Patrick Dempsey, Tom Cruise o Michael Douglas presumen de automóvil http://motor.es.msn.com/coches/galeria.aspx?cp-documentid=152634169-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Music on Hold problema
Any ideas, please? Anahi Ludueña From: a_ludu...@hotmail.com To: asterisk-users@lists.digium.com Date: Thu, 17 Jun 2010 19:54:30 + Subject: Re: [asterisk-users] Music on Hold problema I have wav files in the /var/lib/asterisk/mohmp3... Anahi Ludueña From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Thu, 17 Jun 2010 14:36:00 -0500 Subject: Re: [asterisk-users] Music on Hold problema I see that moh is trying sln format, then ulaw, then failing. Do you have moh files in either of these formats? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Anahi Ludueña Sent: Thursday, June 17, 2010 2:24 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Music on Hold problema Hi people, I have a problem with Music On Hold, it is stopped just after starting... This is the log: [Jun 17 19:20:22] VERBOSE[20784] logger.c: -- Executing [...@test:1] NoOp(SIP/7PBX-08229d18, Start) in new stack [Jun 17 19:20:22] DEBUG[20784] pbx.c: Launching 'MusicOnHold' [Jun 17 19:20:22] VERBOSE[20784] logger.c: -- Executing [...@test:2] MusicOnHold(SIP/7PBX-08229d18, ) in new stack [Jun 17 19:20:22] DEBUG[20784] channel.c: Set channel SIP/7PBX-08229d18 to write format slin [Jun 17 19:20:22] VERBOSE[20784] logger.c: -- Started music on hold, class 'default', on channel 'SIP/7PBX-08229d18' [Jun 17 19:20:22] DEBUG[20784] channel.c: Scheduling timer at 160 sample intervals [Jun 17 19:20:22] DEBUG[20784] channel.c: Auto-deactivating generator [Jun 17 19:20:22] DEBUG[20784] channel.c: Set channel SIP/7PBX-08229d18 to write format ulaw [Jun 17 19:20:22] VERBOSE[20784] logger.c: -- Stopped music on hold on SIP/7PBX-08229d18 [Jun 17 19:20:22] DEBUG[20784] channel.c: Scheduling timer at 0 sample intervals Could you help me with this? Thanks, Anahi Ludueña Disfruta de Hotmail y Messenger en tu móvil con YOIGO. ¡Hazlo ya! ¿Un navegador seguro buscando estás? ¡Protegete ya en www.ayudartepodria.com! _ ¿Quieres descubrir todos los trucos de Windows 7? ¡Hazlo aquí! http://www.sietesunpueblodeexpertos.com/index_windows7.html-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Music on Hold problema
The list of /var/lib/asterisk/mohmp3 is: -rw-rw 4 asterisk asterisk 184 Oct 19 2009 LICENSE-asterisk-moh-freeplay-wav -rw-rw-r-- 4 asterisk asterisk 882748 Oct 19 2009 QuajiroPromo.sln -rw-rw-r-- 4 asterisk asterisk 834682 Oct 19 2009 TristeAlegriaPromo.sln -rw-rw 4 asterisk asterisk 1939794 Oct 19 2009 fpm-calm-river.wav -rw-rw 4 asterisk asterisk 2582196 Oct 19 2009 fpm-sunshine.wav -rw-rw 4 asterisk asterisk 2217318 Oct 19 2009 fpm-world-mix.wav And the musiconhold.conf is: [default] mode=files directory=/var/lib/asterisk/mohmp3 random=yes [none] mode=files directory=/dev/null Thanks, Anahi Ludueña From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Fri, 18 Jun 2010 09:26:16 -0500 Subject: Re: [asterisk-users] Music on Hold problema Post the /var/lib/asterisk/mohmp3 listing and musiconhold.conf From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Anahi Ludueña Sent: Friday, June 18, 2010 9:18 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Music on Hold problema Any ideas, please? Anahi Ludueña From: a_ludu...@hotmail.com To: asterisk-users@lists.digium.com Date: Thu, 17 Jun 2010 19:54:30 + Subject: Re: [asterisk-users] Music on Hold problema I have wav files in the /var/lib/asterisk/mohmp3... Anahi Ludueña From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Thu, 17 Jun 2010 14:36:00 -0500 Subject: Re: [asterisk-users] Music on Hold problema I see that moh is trying sln format, then ulaw, then failing. Do you have moh files in either of these formats? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Anahi Ludueña Sent: Thursday, June 17, 2010 2:24 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Music on Hold problema Hi people, I have a problem with Music On Hold, it is stopped just after starting... This is the log: [Jun 17 19:20:22] VERBOSE[20784] logger.c: -- Executing [...@test:1] NoOp(SIP/7PBX-08229d18, Start) in new stack [Jun 17 19:20:22] DEBUG[20784] pbx.c: Launching 'MusicOnHold' [Jun 17 19:20:22] VERBOSE[20784] logger.c: -- Executing [...@test:2] MusicOnHold(SIP/7PBX-08229d18, ) in new stack [Jun 17 19:20:22] DEBUG[20784] channel.c: Set channel SIP/7PBX-08229d18 to write format slin [Jun 17 19:20:22] VERBOSE[20784] logger.c: -- Started music on hold, class 'default', on channel 'SIP/7PBX-08229d18' [Jun 17 19:20:22] DEBUG[20784] channel.c: Scheduling timer at 160 sample intervals [Jun 17 19:20:22] DEBUG[20784] channel.c: Auto-deactivating generator [Jun 17 19:20:22] DEBUG[20784] channel.c: Set channel SIP/7PBX-08229d18 to write format ulaw [Jun 17 19:20:22] VERBOSE[20784] logger.c: -- Stopped music on hold on SIP/7PBX-08229d18 [Jun 17 19:20:22] DEBUG[20784] channel.c: Scheduling timer at 0 sample intervals Could you help me with this? Thanks, Anahi Ludueña Disfruta de Hotmail y Messenger en tu móvil con YOIGO. ¡Hazlo ya! ¿Un navegador seguro buscando estás? ¡Protegete ya en www.ayudartepodria.com! Del Lado Oscuro de Internet protegerte puedes. ¡Entra ya en www.ayudartepodria.com! _ ¿Quieres descubrir todos los trucos de Windows 7? ¡Hazlo aquí! http://www.sietesunpueblodeexpertos.com/index_windows7.html-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Music on Hold problema
Hi people, I have a problem with Music On Hold, it is stopped just after starting... This is the log: [Jun 17 19:20:22] VERBOSE[20784] logger.c: -- Executing [...@test:1] NoOp(SIP/7PBX-08229d18, Start) in new stack [Jun 17 19:20:22] DEBUG[20784] pbx.c: Launching 'MusicOnHold' [Jun 17 19:20:22] VERBOSE[20784] logger.c: -- Executing [...@test:2] MusicOnHold(SIP/7PBX-08229d18, ) in new stack [Jun 17 19:20:22] DEBUG[20784] channel.c: Set channel SIP/7PBX-08229d18 to write format slin [Jun 17 19:20:22] VERBOSE[20784] logger.c: -- Started music on hold, class 'default', on channel 'SIP/7PBX-08229d18' [Jun 17 19:20:22] DEBUG[20784] channel.c: Scheduling timer at 160 sample intervals [Jun 17 19:20:22] DEBUG[20784] channel.c: Auto-deactivating generator [Jun 17 19:20:22] DEBUG[20784] channel.c: Set channel SIP/7PBX-08229d18 to write format ulaw [Jun 17 19:20:22] VERBOSE[20784] logger.c: -- Stopped music on hold on SIP/7PBX-08229d18 [Jun 17 19:20:22] DEBUG[20784] channel.c: Scheduling timer at 0 sample intervals Could you help me with this? Thanks, Anahi Ludueña _ Sé el protagonista de GQ con Messenger y Vodafone Blackberry. ¡Y gana premios! http://serviciosmoviles.es.msn.com/messenger/vodafone.aspx-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Music on Hold problema
I have wav files in the /var/lib/asterisk/mohmp3... Anahi Ludueña From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Thu, 17 Jun 2010 14:36:00 -0500 Subject: Re: [asterisk-users] Music on Hold problema I see that moh is trying sln format, then ulaw, then failing. Do you have moh files in either of these formats? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Anahi Ludueña Sent: Thursday, June 17, 2010 2:24 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Music on Hold problema Hi people, I have a problem with Music On Hold, it is stopped just after starting... This is the log: [Jun 17 19:20:22] VERBOSE[20784] logger.c: -- Executing [...@test:1] NoOp(SIP/7PBX-08229d18, Start) in new stack [Jun 17 19:20:22] DEBUG[20784] pbx.c: Launching 'MusicOnHold' [Jun 17 19:20:22] VERBOSE[20784] logger.c: -- Executing [...@test:2] MusicOnHold(SIP/7PBX-08229d18, ) in new stack [Jun 17 19:20:22] DEBUG[20784] channel.c: Set channel SIP/7PBX-08229d18 to write format slin [Jun 17 19:20:22] VERBOSE[20784] logger.c: -- Started music on hold, class 'default', on channel 'SIP/7PBX-08229d18' [Jun 17 19:20:22] DEBUG[20784] channel.c: Scheduling timer at 160 sample intervals [Jun 17 19:20:22] DEBUG[20784] channel.c: Auto-deactivating generator [Jun 17 19:20:22] DEBUG[20784] channel.c: Set channel SIP/7PBX-08229d18 to write format ulaw [Jun 17 19:20:22] VERBOSE[20784] logger.c: -- Stopped music on hold on SIP/7PBX-08229d18 [Jun 17 19:20:22] DEBUG[20784] channel.c: Scheduling timer at 0 sample intervals Could you help me with this? Thanks, Anahi Ludueña Disfruta de Hotmail y Messenger en tu móvil con YOIGO. ¡Hazlo ya! _ ¿Un navegador seguro buscando estás? ¡Protegete ya en www.ayudartepodria.com! www.ayudartepodria.com-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call ended after 31 seconds
Yes, I'm using XLite... Anahi Ludueña From: l...@virtutel.ca To: asterisk-users@lists.digium.com Date: Fri, 11 Jun 2010 20:05:39 -0400 Subject: Re: [asterisk-users] Call ended after 31 seconds You`re using Xlite/eyeBeam by any chance? Mike From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Anahi Ludueña Sent: Friday, June 11, 2010 16:12 To: asterisk-users@lists.digium.com Subject: [asterisk-users] Call ended after 31 seconds Hi people, I have a problem with some extensions. The calls are ended after 31/35 seconds, also, it depends on the number which I call. This is the log, but I've not been able to find something wrong... Any ideas? [Jun 11 15:50:46] DEBUG[26071] app_macro.c: Executed application: ExecIf [Jun 11 15:50:46] VERBOSE[26071] logger.c: -- Executing [...@macro-dialout-trunk:16] Macro(SIP/3000-6d07, dialout-trunk-predial-hook|) in new stack [Jun 11 15:50:46] VERBOSE[26071] logger.c: -- Executing [...@macro-dialout-trunk-predial-hook:1] MacroExit(SIP/3000-6d07, ) in new stack [Jun 11 15:50:46] DEBUG[26071] app_macro.c: Executed application: Macro [Jun 11 15:50:46] VERBOSE[26071] logger.c: -- Executing [...@macro-dialout-trunk:17] GotoIf(SIP/3000-6d07, 0?bypass|1) in new stack [Jun 11 15:50:46] DEBUG[26071] app_macro.c: Executed application: GotoIf [Jun 11 15:50:46] VERBOSE[26071] logger.c: -- Executing [...@macro-dialout-trunk:18] GotoIf(SIP/3000-6d07, 0?customtrunk) in new stack [Jun 11 15:50:46] DEBUG[26071] app_macro.c: Executed application: GotoIf [Jun 11 15:50:46] VERBOSE[26071] logger.c: -- Executing [...@macro-dialout-trunk:19] Dial(SIP/3000-6d07, SIP/GAF/|300|) in new stack [Jun 11 15:50:46] NOTICE[26071] app_dial.c: Hey! chan SIP/3000-6d07's context='macro-dialout-trunk', and exten='s' [Jun 11 15:50:46] VERBOSE[26071] logger.c: -- Called SIP/GAF/ [Jun 11 15:50:49] VERBOSE[26071] logger.c: -- SIP/GAF-6 is ringing [Jun 11 15:50:49] VERBOSE[26071] logger.c: -- SIP/GAF-6 is making progress passing it to SIP/3000-6d07 [Jun 11 15:50:56] VERBOSE[26071] logger.c: -- SIP/GAF-6 answered SIP/3000-6d07 [Jun 11 15:50:56] VERBOSE[26071] logger.c: -- Packet2Packet bridging SIP/3000-6d07 and SIP/GAF-6 [Jun 11 15:51:27] VERBOSE[26071] logger.c: -- Executing [...@macro-dialout-trunk:1] Macro(SIP/3000-6d07, hangupcall|) in new stack [Jun 11 15:51:27] VERBOSE[26071] logger.c: -- Executing [...@macro-hangupcall:1] GotoIf(SIP/3000-6d07, 1?skiprg) in new stack [Jun 11 15:51:27] VERBOSE[26071] logger.c: -- Goto (macro-hangupcall,s,4) [Jun 11 15:51:27] DEBUG[26071] app_macro.c: Executed application: GotoIf [Jun 11 15:51:27] VERBOSE[26071] logger.c: -- Executing [...@macro-hangupcall:4] GotoIf(SIP/3000-6d07, 1?skipblkvm) in new stack [Jun 11 15:51:27] VERBOSE[26071] logger.c: -- Goto (macro-hangupcall,s,7) [Jun 11 15:51:27] DEBUG[26071] app_macro.c: Executed application: GotoIf [Jun 11 15:51:27] VERBOSE[26071] logger.c: -- Executing [...@macro-hangupcall:7] GotoIf(SIP/3000-6d07, 1?theend) in new stack [Jun 11 15:51:27] VERBOSE[26071] logger.c: -- Goto (macro-hangupcall,s,9) [Jun 11 15:51:27] DEBUG[26071] app_macro.c: Executed application: GotoIf [Jun 11 15:51:27] VERBOSE[26071] logger.c: -- Executing [...@macro-hangupcall:9] Hangup(SIP/3000-6d07, ) in new stack [Jun 11 15:51:27] VERBOSE[26071] logger.c: == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/3000-6d07' in macro 'hangupcall' [Jun 11 15:51:27] VERBOSE[26071] logger.c: == Spawn h extension (macro-dialout-trunk, h, 1) exited non-zero on 'SIP/3000-6d07' [Jun 11 15:51:27] VERBOSE[26071] logger.c: == Spawn extension (macro-dialout-trunk, s, 19) exited non-zero on 'SIP/3000-6d07' in macro 'dialout-trunk' [Jun 11 15:51:27] VERBOSE[26071] logger.c: == Spawn extension (from-internal, xxx, 5) exited non-zero on 'SIP/3000-6d07' Thanks, Anahi Ludueña Noticias, servicios, tendencias. Haz de MSN.ES tu pág. de inicio _ Sé el protagonista de GQ con Messenger y Vodafone Blackberry. ¡Y gana premios! http://serviciosmoviles.es.msn.com/messenger/vodafone.aspx-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call ended after 31 seconds
Hi people, I have a problem with some extensions. The calls are ended after 31/35 seconds, also, it depends on the number which I call. This is the log, but I've not been able to find something wrong... Any ideas? [Jun 11 15:50:46] DEBUG[26071] app_macro.c: Executed application: ExecIf [Jun 11 15:50:46] VERBOSE[26071] logger.c: -- Executing [...@macro-dialout-trunk:16] Macro(SIP/3000-6d07, dialout-trunk-predial-hook|) in new stack [Jun 11 15:50:46] VERBOSE[26071] logger.c: -- Executing [...@macro-dialout-trunk-predial-hook:1] MacroExit(SIP/3000-6d07, ) in new stack [Jun 11 15:50:46] DEBUG[26071] app_macro.c: Executed application: Macro [Jun 11 15:50:46] VERBOSE[26071] logger.c: -- Executing [...@macro-dialout-trunk:17] GotoIf(SIP/3000-6d07, 0?bypass|1) in new stack [Jun 11 15:50:46] DEBUG[26071] app_macro.c: Executed application: GotoIf [Jun 11 15:50:46] VERBOSE[26071] logger.c: -- Executing [...@macro-dialout-trunk:18] GotoIf(SIP/3000-6d07, 0?customtrunk) in new stack [Jun 11 15:50:46] DEBUG[26071] app_macro.c: Executed application: GotoIf [Jun 11 15:50:46] VERBOSE[26071] logger.c: -- Executing [...@macro-dialout-trunk:19] Dial(SIP/3000-6d07, SIP/GAF/|300|) in new stack [Jun 11 15:50:46] NOTICE[26071] app_dial.c: Hey! chan SIP/3000-6d07's context='macro-dialout-trunk', and exten='s' [Jun 11 15:50:46] VERBOSE[26071] logger.c: -- Called SIP/GAF/ [Jun 11 15:50:49] VERBOSE[26071] logger.c: -- SIP/GAF-6 is ringing [Jun 11 15:50:49] VERBOSE[26071] logger.c: -- SIP/GAF-6 is making progress passing it to SIP/3000-6d07 [Jun 11 15:50:56] VERBOSE[26071] logger.c: -- SIP/GAF-6 answered SIP/3000-6d07 [Jun 11 15:50:56] VERBOSE[26071] logger.c: -- Packet2Packet bridging SIP/3000-6d07 and SIP/GAF-6 [Jun 11 15:51:27] VERBOSE[26071] logger.c: -- Executing [...@macro-dialout-trunk:1] Macro(SIP/3000-6d07, hangupcall|) in new stack [Jun 11 15:51:27] VERBOSE[26071] logger.c: -- Executing [...@macro-hangupcall:1] GotoIf(SIP/3000-6d07, 1?skiprg) in new stack [Jun 11 15:51:27] VERBOSE[26071] logger.c: -- Goto (macro-hangupcall,s,4) [Jun 11 15:51:27] DEBUG[26071] app_macro.c: Executed application: GotoIf [Jun 11 15:51:27] VERBOSE[26071] logger.c: -- Executing [...@macro-hangupcall:4] GotoIf(SIP/3000-6d07, 1?skipblkvm) in new stack [Jun 11 15:51:27] VERBOSE[26071] logger.c: -- Goto (macro-hangupcall,s,7) [Jun 11 15:51:27] DEBUG[26071] app_macro.c: Executed application: GotoIf [Jun 11 15:51:27] VERBOSE[26071] logger.c: -- Executing [...@macro-hangupcall:7] GotoIf(SIP/3000-6d07, 1?theend) in new stack [Jun 11 15:51:27] VERBOSE[26071] logger.c: -- Goto (macro-hangupcall,s,9) [Jun 11 15:51:27] DEBUG[26071] app_macro.c: Executed application: GotoIf [Jun 11 15:51:27] VERBOSE[26071] logger.c: -- Executing [...@macro-hangupcall:9] Hangup(SIP/3000-6d07, ) in new stack [Jun 11 15:51:27] VERBOSE[26071] logger.c: == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/3000-6d07' in macro 'hangupcall' [Jun 11 15:51:27] VERBOSE[26071] logger.c: == Spawn h extension (macro-dialout-trunk, h, 1) exited non-zero on 'SIP/3000-6d07' [Jun 11 15:51:27] VERBOSE[26071] logger.c: == Spawn extension (macro-dialout-trunk, s, 19) exited non-zero on 'SIP/3000-6d07' in macro 'dialout-trunk' [Jun 11 15:51:27] VERBOSE[26071] logger.c: == Spawn extension (from-internal, xxx, 5) exited non-zero on 'SIP/3000-6d07' Thanks, Anahi Ludueña _ Los cochazos de los famosos Patrick Dempsey, Tom Cruise o Michael Douglas presumen de automóvil http://motor.es.msn.com/coches/galeria.aspx?cp-documentid=152634169-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Press twice *
Hi people, I need to detect when the user presses twice *... In the dialplan I added the following, but it doesn't work. Could you help me with that? exten = **,1,. Anahi Ludueña _ ¿Un navegador seguro buscando estás? ¡Protegete ya en www.ayudartepodria.com! www.ayudartepodria.com-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dial an extension with follow me
Hi people, I have an extension which has configured the follow me (it derives to an IVR). If in my dialplan I put Dial(extenX) (where extenX is that extension) and if it is not available, it should execute the IVR, is that right? Well, I think it should be, but it doesn't... Here is my CLI: Starting SIP/CALLUS-0b3f at join-dial,,1 failed so falling back to exten 's' -- Executing [...@join-dial:1] NoOp(SIP/CALLUS-0b3f, join-dial: START) in new stack -- Executing [...@join-dial:2] GotoIf(SIP/CALLUS-0b3f, 1?queue:conti) in new stack -- Goto (join-dial,s,3) -- Executing [...@join-dial:3] Gosub(SIP/CALLUS-0b3f, call-fm|s|1) in new stack -- Executing [...@call-fm:1] NoOp(SIP/CALLUS-0b3f, Start) in new stack -- Executing [...@call-fm:2] Dial(SIP/CALLUS-0b3f, SIP/3006) in new stack == Everyone is busy/congested at this time (1:0/0/1) == Auto fallthrough, channel 'SIP/CALLUS-0b3f' status is 'CHANUNAVAIL' Thanks, Anahi Ludueña _ Aprende los trucos de Windows 7 con la gente que ya lo han probado Windows 7. http://www.sietesunpueblodeexpertos.com/index_windows7.html-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Playback in h extension
Hi people, I'm trying to execute the PlayBack command in the h extension... but it is not played... is it possible to do that?Thanks, Anahi Anahi Ludueña _ Ahora Messenger en tu Blackberry® 8520 con Movistar por 0 €. ¿A qué esperas? http://serviciosmoviles.es.msn.com/messenger/blackberry.aspx-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AMI logs
Hi, I'm executing some commands using AMI... I suppose the log is saved in some place, but I don't know where... where is it saved?More details: I'm executing a UpdateConfig in the voicemail.conf file, but the file is not updated, so I would like to know why...Thanks, Anahi Anahi Ludueña _ Ahora Messenger en tu Blackberry® 8520 con Movistar por 0 €. ¿A qué esperas? http://serviciosmoviles.es.msn.com/messenger/blackberry.aspx-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Manager Logged off
Hi People, I don't know if my problem should be reported in this forum, but maybe somebody knows about it. I'm using the tool .NET WebService Studio to test the web service which is working with asterisk by AMI. It is working fine, the dialplan is executed correctly... the problem is when the web service is consumed by my program (Genexus). I've been checking the log and the differences when I use .NET Webservice Studio and when I use my program... The difference I found is that the Manager is logged off and that is the reason why the dialplan is not executed fine in the second case. [Feb 24 08:08:20] DEBUG[1212] app_meetme.c: Cmdline: 7|k|1 [Feb 24 08:08:20] VERBOSE[1212] logger.c: == Manager 'asteriskWS' logged off from 67.63.42.120 The previous log shows that DEBUG line which is not shown in the log when I use the .NET Webservice Studio. Anybody knows why the Manager is logged off? Thanks, _ ¿Aún no sabes qué móvil eres? ¡Descúbrelo aquí! http://www.quemovileres.com/-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Voicemail after hangup
Hi people, just a question: Is it possible to execute Voicemail command in the h extension? (after hangup the channel). Because if I put it before it, it works right, but if I put it there, it doesn't... The log is: -- Executing [...@cont-mine:1] NoOp(SIP/3005-096736a8, End of cont-mine) in new stack -- Executing [...@cont-mine:2] VoiceMail(SIP/3005-096736a8, 3003|su) in new stack -- SIP/3005-096736a8 Playing '/var/spool/asterisk/voicemail/default/3003/unavail' (language '') == Spawn extension (cont-mine, h, 2) exited non-zero on 'SIP/3005-096736a8' Thanks, _ Chatea sin límites en Messenger con la tarifa plana de Orange http://serviciosmoviles.es.msn.com/messenger/orange.aspx___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Hangup
Hi, is it possible to hangup a channel from another channel? I want to finish a call from another channel, but if I put exten = h,n,HangUp(channelname) it doesn't hangup... Is that correct? Thanks, _ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hangup, SoftHangup
Thanks Phillipp!, it works! Anahi Ludueña Date: Tue, 10 Nov 2009 14:44:09 +0100 From: philipp.kemp...@amooma.de To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Hangup, SoftHangup Anahi Ludueña schrieb: is it possible to hangup a channel from another channel? I want to finish a call from another channel, but if I put exten = h,n,HangUp(channelname) it doesn't hangup... Is that correct? You need to use the SoftHangup() application. core show application SoftHangup Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Sólo hay un loro experto en Windows 7 en todo el mundo. Y vive en Sietes ¡Cónocelo! http://www.sietesunpueblodeexpertos.com/___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Voicemail file
Hi all, When somebody leaves a message in the voicemailbox, is there a way to know the file name of it? I need to return the voicemail file name in the deadagi command. Thanks, Anahi _ Convierte las fotos que más te gustan en tu nuevo fondo de escritorio para el ordenador. Es fácil y además gratis http://wallpapers.msn.com/es-es___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail file
Yes, but is there a way to know the filename of that message? For example: msg0029.wav? I know where it is saved, but if I want to return it, I need to find the last one... and it is not recommended in my opinion... Thanks, Anahi Ludueña From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Fri, 30 Oct 2009 10:14:27 -0500 Subject: Re: [asterisk-users] Voicemail file Assuming you aren’t writing your VM’s to a database, the voicemail will be in /var/spool/asterisk/voicemail/default/extension/INBOX/msg.WAV (wav, gsm, txt – depends on voicemail.conf). From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Anahi Ludueña Sent: Friday, October 30, 2009 10:06 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Voicemail file Hi all, When somebody leaves a message in the voicemailbox, is there a way to know the file name of it? I need to return the voicemail file name in the deadagi command. Thanks, Anahi Entra al Nuevo Canal Motor y descubre por qué los coches más rápidos sólo aparcan en MSN. Nuevo diseño, más completo y abierto a tu opinión. ¡Nuevo Canal Motor! _ Infórmate, mantente en contacto y encuéntralo todo, a la vez. Con la nueva Toolbar de MSN nunca has tenido tantas ventajas en tan poco espacio. http://toolbar.es.msn.com/___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail file
Thanks people,
I've already found the way...
The variable ${VM_MESSAGEFILE} contains what I need...
Bye,
Anahi Ludueña
From: a_ludu...@hotmail.com
To: asterisk-users@lists.digium.com
Date: Fri, 30 Oct 2009 15:18:25 +
Subject: Re: [asterisk-users] Voicemail file
Yes, but is there a way to know the filename of that message?
For example: msg0029.wav?
I know where it is saved, but if I want to return it, I need to find the last
one... and it is not recommended in my opinion...
Thanks,
Anahi Ludueña
From: da...@debsinc.com
To: asterisk-users@lists.digium.com
Date: Fri, 30 Oct 2009 10:14:27 -0500
Subject: Re: [asterisk-users] Voicemail file
Assuming you aren’t writing your VM’s
to a database, the voicemail will be in
/var/spool/asterisk/voicemail/default/extension/INBOX/msg.WAV
(wav, gsm, txt – depends on voicemail.conf).
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Anahi Ludueña
Sent: Friday, October 30, 2009
10:06 AM
To:
asterisk-users@lists.digium.com
Subject: [asterisk-users]
Voicemail file
Hi all,
When somebody leaves a message in the voicemailbox, is there a way to know the
file name of it?
I need to return the voicemail file name in the deadagi command.
Thanks,
Anahi
Entra al Nuevo Canal Motor y descubre por qué los coches
más rápidos sólo aparcan en MSN. Nuevo diseño, más completo y abierto a tu
opinión. ¡Nuevo Canal Motor!
Todo el espacio y cuidado que merecen tus fotos digitales lo tienes en Windows
Live Fotos. ¡Pruébalo!
_
Infórmate, mantente en contacto y encuéntralo todo, a la vez. Con la nueva
Toolbar de MSN nunca has tenido tantas ventajas en tan poco espacio.
http://toolbar.es.msn.com/___
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Re: [asterisk-users] R: R: R: CDR(billsec)
Thanks Matt! It works now! Bye... Anahi Ludueña Date: Fri, 30 Oct 2009 01:20:02 +1300 From: li...@venturevoip.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] R: R: R: CDR(billsec) On 30/10/09 1:10 AM, Alexandru Oniciuc wrote: Thank you! My bad,the CDR function was working on 1.4, I can confirm that endbeforehexten=yes does the trick, I've just tried it :] WOW: On 29/10/09 5:56 AM, Alexandru Oniciuc wrote: 12h difference! :D Yeah based in New Zealand - we're just about ahead of everybody - in fact it's 1:20 in the morning so I probably should go to sleep :) -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ ¿Sabías que ahora puedes hablar por Messenger desde Hotmail con todos tus contactos? Revisa tu correo mientras conversas con tus amigos. http://www.hotmail.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CDR(billsec)
Hi people, when I try to get the billsec in the dialplan, it is 0... but if
after that I check the database, it is right (not 0).
I'm trying to get it in the h extension, like:
exten = h,1,Noop(End)
exten = h,n,Noop(Time is ${CDR(billsec)})
Is it updated after the extension h is executed? In that case, how can I get
the call duration in the h extension?
Thanks,
Anahi Ludueña
_
Infórmate, mantente en contacto y encuéntralo todo, a la vez. Con la nueva
Toolbar de MSN nunca has tenido tantas ventajas en tan poco espacio.
http://toolbar.es.msn.com/___
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Re: [asterisk-users] Dial a external number with extension
Thanks, I need to make a conference between 2 numbers, one of them is external and it has an extension. So, I need to dial the number and later enter the extension, how can I do that? _ ¿Sabías que ahora puedes hablar por Messenger desde Hotmail con todos tus contactos? Revisa tu correo mientras conversas con tus amigos. http://www.hotmail.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dial a external number with extension
Hi People, I need to dial an external number, when it is answered, I should digit the extension. How can I do that in the DialPlan? Thanks, Anahi Ludueña _ ¿Sabías que ahora puedes hablar por Messenger desde Hotmail con todos tus contactos? Revisa tu correo mientras conversas con tus amigos. http://www.hotmail.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dialplan problem
Hi people, I have the following dialplan, but it doesn't have the behavior that I think it should have. [default] exten = 2001,1,Answer exten = 2001,n,Dial(local/3005) exten = 2001,n,Hangup exten = 3005,1,Set(__RINGTIMER=10) exten = 3005,n,Macro(exten-vm,novm,3005) exten = 3005,n,Hangup When I execute the Originate (AMI) with the argument Channel=local/2001, It rings the local/3005 but 2 calls are generated... Why? Or how should the dialplan be to generate only one call on the local/3005? Thanks, The log is: -- Executing [2...@default:1] Answer(Local/2...@default-b60f,2, ) in new stack == Manager 'asteriskWS' logged off from ... -- Executing [2...@default:2] Dial(Local/2...@default-b60f,2, local/3005) in new stack -- Called 3005 -- Executing [2...@default:1] Answer(Local/2...@default-b60f,1, ) in new stack -- Executing [2...@default:2] Dial(Local/2...@default-b60f,1, local/3005) in new stack -- Called 3005 -- Executing [3...@default:1] Set(Local/3...@default-6cc4,2, __RINGTIMER=10) in new stack -- Executing [3...@default:2] Macro(Local/3...@default-6cc4,2, exten-vm|novm|3005) in new stack -- Executing [...@macro-exten-vm:1] Macro(Local/3...@default-6cc4,2, user-callerid) in new stack -- Executing [...@macro-user-callerid:1] NoOp(Local/3...@default-6cc4,2, user-callerid: ) in new stack -- Executing [...@macro-user-callerid:2] Set(Local/3...@default-6cc4,2, AMPUSER=) in new stack -- Executing [...@macro-user-callerid:3] GotoIf(Local/3...@default-6cc4,2, 1?report) in new stack -- Goto (macro-user-callerid,s,13) -- Executing [...@macro-user-callerid:13] NoOp(Local/3...@default-6cc4,2, TTL: ARG1: novm) in new stack -- Executing [...@macro-user-callerid:14] GotoIf(Local/3...@default-6cc4,2, 0?continue) in new stack -- Executing [...@macro-user-callerid:15] Set(Local/3...@default-6cc4,2, __TTL=64) in new stack -- Executing [...@macro-user-callerid:16] GotoIf(Local/3...@default-6cc4,2, 1?continue) in new stack -- Goto (macro-user-callerid,s,23) -- Executing [...@macro-user-callerid:23] NoOp(Local/3...@default-6cc4,2, Using CallerID ) in new stack -- Executing [...@macro-exten-vm:2] Set(Local/3...@default-6cc4,2, RingGroupMethod=none) in new stack -- Executing [...@macro-exten-vm:3] Set(Local/3...@default-6cc4,2, VMBOX=novm) in new stack -- Executing [...@macro-exten-vm:4] Set(Local/3...@default-6cc4,2, EXTTOCALL=3005) in new stack -- Executing [...@macro-exten-vm:5] Set(Local/3...@default-6cc4,2, CFUEXT=) in new stack -- Executing [...@macro-exten-vm:6] Set(Local/3...@default-6cc4,2, CFBEXT=) in new stack -- Executing [...@macro-exten-vm:7] Set(Local/3...@default-6cc4,2, RT=) in new stack -- Executing [...@macro-exten-vm:8] Macro(Local/3...@default-6cc4,2, record-enable|3005|IN) in new stack -- Executing [...@macro-record-enable:1] GotoIf(Local/3...@default-6cc4,2, 0?2:4) in new stack -- Goto (macro-record-enable,s,4) -- Executing [...@macro-record-enable:4] AGI(Local/3...@default-6cc4,2, recordingcheck|20091008-093826|1255009106.184) in new stack -- Executing [3...@default:1] Set(Local/3...@default-e393,2, __RINGTIMER=10) in new stack -- Executing [3...@default:2] Macro(Local/3...@default-e393,2, exten-vm|novm|3005) in new stack -- Executing [...@macro-exten-vm:1] Macro(Local/3...@default-e393,2, user-callerid) in new stack -- Executing [...@macro-user-callerid:1] NoOp(Local/3...@default-e393,2, user-callerid: ) in new stack -- Executing [...@macro-user-callerid:2] Set(Local/3...@default-e393,2, AMPUSER=) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck -- Executing [...@macro-user-callerid:3] GotoIf(Local/3...@default-e393,2, 1?report) in new stack -- Goto (macro-user-callerid,s,13) -- Executing [...@macro-user-callerid:13] NoOp(Local/3...@default-e393,2, TTL: ARG1: novm) in new stack -- Executing [...@macro-user-callerid:14] GotoIf(Local/3...@default-e393,2, 0?continue) in new stack -- Executing [...@macro-user-callerid:15] Set(Local/3...@default-e393,2, __TTL=64) in new stack -- Executing [...@macro-user-callerid:16] GotoIf(Local/3...@default-e393,2, 1?continue) in new stack -- Goto (macro-user-callerid,s,23) -- Executing [...@macro-user-callerid:23] NoOp(Local/3...@default-e393,2, Using CallerID ) in new stack -- Executing [...@macro-exten-vm:2] Set(Local/3...@default-e393,2, RingGroupMethod=none) in new stack -- Executing [...@macro-exten-vm:3] Set(Local/3...@default-e393,2, VMBOX=novm) in new stack -- Executing [...@macro-exten-vm:4] Set(Local/3...@default-e393,2, EXTTOCALL=3005) in new stack -- Executing [...@macro-exten-vm:5] Set(Local/3...@default-e393,2, CFUEXT=) in new stack -- Executing [...@macro-exten-vm:6] Set(Local/3...@default-e393,2, CFBEXT=) in new stack -- Executing [...@macro-exten-vm:7]
Re: [asterisk-users] Dialplan problem
Thanks, the answers helped me... I was thinking to execute a macro or another context which performs a DIAL command to a particular number. First I checked how it was working doing DIAL directly... that is the reason why I put that context. Thanks again... Anahi Ludueña Date: Fri, 9 Oct 2009 00:26:36 +0200 From: i...@albafotonica.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Dialplan problem Anahi Ludueña wrote: Hi people, I have the following dialplan, but it doesn't have the behavior that I think it should have. [default] exten = 2001,1,Answer exten = 2001,n,Dial(local/3005) exten = 2001,n,Hangup exten = 3005,1,Set(__RINGTIMER=10) exten = 3005,n,Macro(exten-vm,novm,3005) exten = 3005,n,Hangup Why would you do that anyway? use 'Goto(3005,1)' instead of 'Dial' Saludos -- Iván Stepaniuk Alba Fotónica S.L. www.albafotonica.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Convierte las fotos que más te gustan en tu nuevo fondo de escritorio para el ordenador. Es fácil y además gratis http://wallpapers.msn.com/es-es___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] OriginateResponse Event
Hi people, I'm executing some parallel Originate async, is there a way to know the result of each originate?... I was looking at the OriginateResponse event, but I don't know how to get it from my web service. Also, if I have 3 originate in parallel, how can I identify the OriginateResponse of each one? Thanks in advance... Anahi Ludueña _ Nuevo Windows Live, un mundo lleno de posibilidades. Descúbrelo. http://www.microsoft.com/windows/windowslive/default.aspx___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OriginateResponse Event
Thanks Danny, but how can I get it from my web service? Anahi Ludueña From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Mon, 5 Oct 2009 10:03:41 -0500 Subject: Re: [asterisk-users] OriginateResponse Event Each response set has a uniqueid field that designates the start time and call sequence of the call. Unless you manage to start 36K calls simultaneously, you can track each call with this. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Anahi Ludueña Sent: Monday, October 05, 2009 9:56 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] OriginateResponse Event Hi people, I'm executing some parallel Originate async, is there a way to know the result of each originate?... I was looking at the OriginateResponse event, but I don't know how to get it from my web service. Also, if I have 3 originate in parallel, how can I identify the OriginateResponse of each one? Thanks in advance... Anahi Ludueña Nuevo Windows Live, un mundo lleno de posibilidades Descúbrelo. _ Nuevo Windows Live, un mundo lleno de posibilidades. Descúbrelo. http://www.microsoft.com/windows/windowslive/default.aspx___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Followme
Hi everybody, What I need to do is to run a context where I'll pass some phones (for example: 3 numbers). I need to make something like a followme, if the first phone is not answered, I'll call the second one, and so on. That dial plan is not the problem, my problem is when I execute the AMI, I'm using the Originate. It needs a channel as an argument, so the context can be executed; but what channel should I pass there? (the phone numbers are in the Variable argument) Thanks in advance. Anahi Ludueña _ ¿Quieres ver los mejores videos de MSN? Enciende Messenger TV http://messengertv.msn.com/mkt/es-es/default.htm___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Followme
Thanks Danny, It seems I'm doing something wrong. Let forget the followme, I have this context: [new-context] exten = 1,1,Answer exten = 1,2,Dial(sip/1000) exten = 1,3,Playback(sorrynoanswer) exten = 1,4,Hangup Now, I execute the Originate with these parameters: Channel: Local/1 Context: new-context Priority: 1 But it gives this error: Response: Error Message: Originate failed Do you know if there is something wrong? Thanks again. From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Fri, 2 Oct 2009 08:25:58 -0500 Subject: Re: [asterisk-users] Followme Local/1 will run the context without tying up resources. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Anahi Ludueña Sent: Friday, October 02, 2009 8:20 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Followme Hi everybody, What I need to do is to run a context where I'll pass some phones (for example: 3 numbers). I need to make something like a followme, if the first phone is not answered, I'll call the second one, and so on. That dial plan is not the problem, my problem is when I execute the AMI, I'm using the Originate. It needs a channel as an argument, so the context can be executed; but what channel should I pass there? (the phone numbers are in the Variable argument) Thanks in advance. Anahi Ludueña Comparte tus fotos con tus amigos. Más fácil con Windows Live _ Descubre todas las formas en que puedes estar en contacto con amigos y familiares. http://www.microsoft.com/windows/windowslive/default.aspx___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Followme
Thanks, anyway the result is the same... Response: Error Message: Originate failed Anahi Ludueña From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Fri, 2 Oct 2009 10:01:05 -0500 Subject: Re: [asterisk-users] Followme Change the 1’s to s. The 1 assumes that you pressed 1 from an IVR/DTMF selection. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Anahi Ludueña Sent: Friday, October 02, 2009 9:53 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Followme Thanks Danny, It seems I'm doing something wrong. Let forget the followme, I have this context: [new-context] exten = 1,1,Answer exten = 1,2,Dial(sip/1000) exten = 1,3,Playback(sorrynoanswer) exten = 1,4,Hangup Now, I execute the Originate with these parameters: Channel: Local/1 Context: new-context Priority: 1 But it gives this error: Response: Error Message: Originate failed Do you know if there is something wrong? Thanks again. From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Fri, 2 Oct 2009 08:25:58 -0500 Subject: Re: [asterisk-users] Followme Local/1 will run the context without tying up resources. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Anahi Ludueña Sent: Friday, October 02, 2009 8:20 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Followme Hi everybody, What I need to do is to run a context where I'll pass some phones (for example: 3 numbers). I need to make something like a followme, if the first phone is not answered, I'll call the second one, and so on. That dial plan is not the problem, my problem is when I execute the AMI, I'm using the Originate. It needs a channel as an argument, so the context can be executed; but what channel should I pass there? (the phone numbers are in the Variable argument) Thanks in advance. Anahi Ludueña Comparte tus fotos con tus amigos. Más fácil con Windows Live Diferentes formas de estar en contacto con amigos y familiares. Descúbrelas. Descúbrelas. _ Llévate Messenger en el móvil a todas partes ¡Conéctate! http://www.microsoft.com/spain/windowsmobile/messenger/default.mspx___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Followme
Maybe there is another problem. I changed the context like you said. Where is the local channel configured? or is it implicit? Sorry but I'm newbie with Asterisk... Anahi Ludueña From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Fri, 2 Oct 2009 10:15:45 -0500 Subject: Re: [asterisk-users] Followme The 1 (1,1; 1,2; 1,3) needs to be s (s,1; s,2; s,3). It works for me with that change From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Anahi Ludueña Sent: Friday, October 02, 2009 9:53 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Followme Thanks Danny, It seems I'm doing something wrong. Let forget the followme, I have this context: [new-context] exten = 1,1,Answer exten = 1,2,Dial(sip/1000) exten = 1,3,Playback(sorrynoanswer) exten = 1,4,Hangup Now, I execute the Originate with these parameters: Channel: Local/1 Context: new-context Priority: 1 But it gives this error: Response: Error Message: Originate failed Do you know if there is something wrong? Thanks again. From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Fri, 2 Oct 2009 08:25:58 -0500 Subject: Re: [asterisk-users] Followme Local/1 will run the context without tying up resources. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Anahi Ludueña Sent: Friday, October 02, 2009 8:20 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Followme Hi everybody, What I need to do is to run a context where I'll pass some phones (for example: 3 numbers). I need to make something like a followme, if the first phone is not answered, I'll call the second one, and so on. That dial plan is not the problem, my problem is when I execute the AMI, I'm using the Originate. It needs a channel as an argument, so the context can be executed; but what channel should I pass there? (the phone numbers are in the Variable argument) Thanks in advance. Anahi Ludueña Comparte tus fotos con tus amigos. Más fácil con Windows Live Diferentes formas de estar en contacto con amigos y familiares. Descúbrelas. Descúbrelas. _ ¿Quieres ver los mejores videos de MSN? Enciende Messenger TV http://messengertv.msn.com/mkt/es-es/default.htm___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to finish a Meetme
Hi people, I want to make a meetme between 2 numbers. First I enter the number1 into the meetme. It is waiting for the other number, but the other number never entered, so, how can I finish the meetme from the dialplan?. Is it posible by using MeetmeAdmin and kick all the users? Thanks, Anahi Ludueña _ Descubre todas las formas en que puedes estar en contacto con amigos y familiares. http://www.microsoft.com/windows/windowslive/default.aspx___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] UpdateConfig
Thanks, It worked, it seems there was something wrong. The following is working now: Action: UpdateConfig srcFileName: voicemail.conf dstFileName: voicemail.conf Action-00: Append Cat-00: default Var-00: 2000 Value-00: ,Jhon ActionID: 1234 Bye, Anahi Ludueña Date: Tue, 29 Sep 2009 17:50:05 -0500 From: jsm...@digium.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] UpdateConfig - Danny Nicholas da...@debsinc.com wrote: Two questions: 1. do you need an ActionID line? Danny, It's *always* considered best practice to have an ActionID line in AMI commands, so that you can easily differentiate the responses, especially to asynchronous commands. -- Jared Smith Training Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Chatea sin límites en Messenger con la tarifa plana de Orange http://serviciosmoviles.es.msn.com/messenger/orange.aspx___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] UpdateConfig
Hi people, I need to update the voicemail.conf from the UpdateConfig Action (AMI). The problem is that I executed: Action: UpdateConfig srcFileName: voicemail.conf dstFileName: voicemail.conf Action-00:append Cat-00:test Var-00:exten Value-00:999,test But I don't see the changes in the file. Can anybody tell me if there is something wrong in that code? Thanks, Anahi Ludueña _ Descubre todas las formas en que puedes estar en contacto con amigos y familiares. http://www.microsoft.com/windows/windowslive/default.aspx___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] UpdateConfig
Thanks, the result was: Response: Success Anahi Ludueña From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Tue, 29 Sep 2009 15:16:52 -0500 Subject: Re: [asterisk-users] UpdateConfig Two questions: 1. do you need an ActionID line? 2. did you try this in a telnet session so you could see the feedback? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Anahi Ludueña Sent: Tuesday, September 29, 2009 3:08 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] UpdateConfig Hi people, I need to update the voicemail.conf from the UpdateConfig Action (AMI). The problem is that I executed: Action: UpdateConfig srcFileName: voicemail.conf dstFileName: voicemail.conf Action-00:append Cat-00:test Var-00:exten Value-00:999,test But I don't see the changes in the file. Can anybody tell me if there is something wrong in that code? Thanks, Anahi Ludueña Diferentes formas de estar en contacto con amigos y familiares. Descúbrelas. Descúbrelas. _ Descubre todas las formas en que puedes estar en contacto con amigos y familiares. http://www.microsoft.com/windows/windowslive/default.aspx___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MeetMe in Macro
Hi Juan, I didn't use the GoSub application, I put the name of the context in the Originate and the variables and their values in the Variable field. See http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+Originate. Good luck! Anahi Ludueña From: jcard...@tpmex.com To: asterisk-users@lists.digium.com Date: Wed, 23 Sep 2009 10:09:52 -0500 Subject: Re: [asterisk-users] MeetMe in Macro I need the same information, did you find that information Anahi??? Best regards Juan Cardoza De: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] En nombre de Anahi Ludueña Enviado el: Miércoles, 16 de Septiembre de 2009 09:49 a.m. Para: asterisk-users@lists.digium.com Asunto: [asterisk-users] MeetMe in Macro Thanks Miguel, It was my mistake. So, my question is: if I want to call the GoSub application from the Originate Action (using AMI), what I need to put in the context parameter? The GoSub will jump to a special context. Thanks, Date: Wed, 16 Sep 2009 09:34:31 -0500 From: mmol...@millenium.com.co To: asterisk-...@lists.digium.com; asterisk-users@lists.digium.com Subject: Re: [asterisk-dev] MeetMe in Macro Hi, I didn't notice on my first answer, but we are on the -dev list and this is not related to asterisk code developing. I will answer you on the -users list, so we can continue the discussion there. Cheers, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center ¿Quieres que tus amigos de Messenger sigan tus movimientos de Facebook? ¡Conéctalos ya! Teleperformance values: Integrity - Respect - Professionalism - Innovation – Commitment The information contained in this communication is privileged and confidential. The content is intended only for the use of the individual or entity named above. If the reader of this message is not the intended recipient, you are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited. If you have received this communication in error, please notify me immediately by telephone or e-mail, and delete this message from your systems. Please consider the environmental impact of needlessly printing this e-mail. _ Descubre todas las formas en que puedes estar en contacto con amigos y familiares. http://www.microsoft.com/windows/windowslive/default.aspx___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MySQL cmd
Hi people, I'm trying to retrieve data from the database (server MySQL).
I have the following dial plan:
exten = s,1,Noop(Start)
exten = s,n,MYSQL(Connect connid localhost user pass asteriskcdrdb)
exten = s,n,Noop(Connid: ${connid})
...
The problem is that the 3º line is not showing the connid. How can I know the
error?
Thanks,
_
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Re: [asterisk-users] DeadAgi
Thanks guys, I'll take it into account!... Anahi Ludueña Date: Fri, 18 Sep 2009 10:13:12 +0100 From: i...@pack-net.co.uk To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] DeadAgi Steve Edwards wrote: On Thu, 17 Sep 2009, Anahi Ludue?a wrote: Thanks for the answers! The file didn't have the first line! #!/usr/bin/php Glad you found the answer. However... The command ls -l returns: -rwxrwxrwx 1 root root 140 Sep 17 15:42 finconf.php Having an executable with 777 permissions is a very bad idea. Think about somebody (or some program) executing something like: echo rm -f -r /whatever-they-want \ /var/lib/asterisk/agi-bin/finconf.php Agreeing with the above here, really you want the script owned by asterisk.asterisk and permissions of 0755 Ish ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Chatea sin límites en Messenger con la tarifa plana de Orange http://serviciosmoviles.es.msn.com/messenger/orange.aspx___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Audio Files
Hi people, What can I use to transfer the audio files to and from Asterisk? I was searching and I found the following commands: PUT SOUNDFILE and GET SOUNDFILE They are new commands of AGI, but is there another way to do that? Thanks, Anahi Ludueña _ Descubre todas las formas en que puedes estar en contacto con amigos y familiares. http://www.microsoft.com/windows/windowslive/default.aspx___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DeadAgi
Hi people, I have the following dialplan:
[context]
exten = s,1,Noop(Start)
...
exten = h,1,Noop(Ending)
exten = h,n,DEADAGI(finconf.php,${ARG1},${ARG2})
When it is running, the asterisk gives the following error:
-- Launched AGI Script /var/lib/asterisk/agi-bin/finconf.php
== finconf.php|800|: Failed to execute
'/var/lib/asterisk/agi-bin/finconf.php': No such file or directory
But the file is there. The command ls -l returns:
-rwxrwxrwx 1 root root 140 Sep 17 15:42 finconf.php
Why does it return the error?
Thanks,
Anahi Ludueña
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Re: [asterisk-users] DeadAgi
Thanks for the answers! The file didn't have the first line! #!/usr/bin/phpBye! Anahi Ludueña From: tles...@digium.com To: asterisk-users@lists.digium.com Date: Thu, 17 Sep 2009 15:59:21 -0500 Subject: Re: [asterisk-users] DeadAgi On Thursday 17 September 2009 15:06:28 Geraint Lee wrote: 1) does the file exist 2) is it chmod'd to 755 (not sure if this matters though) 3) do you have something like #!/usr/bin/php at the start of the php file? 4) Is the file in MS-DOS format (i.e. do you have \r\n at the end of every line, instead of only \n)? That invisible character (\r) will prevent the file from executing, as Unix is looking for a file on the filesystem named /usr/bin/php\r, and that file probably doesn't exist. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Nuevo Windows Live, un mundo lleno de posibilidades. Descúbrelo. http://www.microsoft.com/windows/windowslive/default.aspx___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MeetMe in Macro
Thanks Miguel, It was my mistake. So, my question is: if I want to call the GoSub application from the Originate Action (using AMI), what I need to put in the context parameter? The GoSub will jump to a special context. Thanks, Date: Wed, 16 Sep 2009 09:34:31 -0500 From: mmol...@millenium.com.co To: asterisk-...@lists.digium.com; asterisk-users@lists.digium.com Subject: Re: [asterisk-dev] MeetMe in Macro Hi, I didn't notice on my first answer, but we are on the -dev list and this is not related to asterisk code developing. I will answer you on the -users list, so we can continue the discussion there. Cheers, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center _ Hay tantos ordenadores como personas. ¡Descubre ahora cuál eres tú! http://www.quepceres.com/___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MeetMe in Macro
Thanks, I asked you to execute the GoSub from the Originate action, because I need to pass some parameters. First, I created a macro since I could pass the parameters from originate. But the macro's problem is it doesn't jump to the particular extension (for example: h extension). So, when you told me that GoSub could replace the Macro, I thought it could be called from the Originate... Do you know if there is another way to pass some parameters to a context from the Originate? Thank you! Anahi Ludueña Date: Wed, 16 Sep 2009 10:27:26 -0500 From: mmol...@millenium.com.co To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] MeetMe in Macro Hi, The GoSub() application is intended for use in the dialplan, not to call it from a Originate Action. What is your specific need? You can Originate to a extension instead of an application an then if you need to execute a subroutine, you can use GoSub() and Return() then you need to on the called context. You can check http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Gosub but the example using the same context is not very clear. A better example would be this: [incoming] exten = s,1,Answer() exten = s,n,Noop(one) exten = s,n,Noop(two) exten = s,n,GoSub(mysub,s,1) exten = s,n,Noop(I returned!) exten = s,n,Hangup [mysub] exten = s,1,Noop(So I'm at a subroutine) exten = s,n,Noop(I need to do special steps) exten = s,n,Playback(tt-monkeys) exten = s,n,Return() Cheers, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center Anahi Ludueña escribió: Thanks Miguel, It was my mistake. So, my question is: if I want to call the GoSub application from the Originate Action (using AMI), what I need to put in the context parameter? The GoSub will jump to a special context. Thanks, Date: Wed, 16 Sep 2009 09:34:31 -0500 From: mmol...@millenium.com.co To: asterisk-...@lists.digium.com; asterisk-users@lists.digium.com Subject: Re: [asterisk-dev] MeetMe in Macro Hi, I didn't notice on my first answer, but we are on the -dev list and this is not related to asterisk code developing. I will answer you on the -users list, so we can continue the discussion there. Cheers, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center _ Hay tantos ordenadores como personas. ¡Descubre ahora cuál eres tú! http://www.quepceres.com/___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Meetme feature
Hi People, I want to do the following steps: - Create a meetme between 2 persons. - First, 1 person (user1) is entered into the meetme. - Second, user2 is entered into the meetme. User2 is the marked user and also he is able to exit the conference by pressing #. - If user2 exited by pressing #, I want the user1 would be able to save a voicemail to the user2. How can I know if the user2 exited the conference by pressing # ? Thanks a lot, bye! _ Comparte tus mejores momentos del verano ¡Hazlo con Windows Live Fotos! http://www.vivelive.com/compartirfotos___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
