search for: touch_monitor

TOUCH_MONITOR is the variable to set if I need to specify my own options for 'One Touch Record' (filetype|filename|m). I cannot get it to work. Can you help? Thanks in advance -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/...

...t; (i.e. "wav:monitor"), res_monitor will prefix the monitored file name with <string> and append that resulting string to a CDRs userfiled all by itself. So basicaly for INCOMING calls I do something like this: exten => 100,1,Set(DYNAMIC_FETAURES=automon) exten => 100,n,Set(TOUCH_MONITOR_FORMAT=wav:monitor) exten => 100,n,Set(TOUCH_MONITOR=${CALLERID}-${EXTEN}) exten => 100,n,Dial(SIP/100,,tw) and in that case (of incoming calls) everything is great and my CDR has userfiled set to something like monitor/auto-{UNIXTIMESTAMP}-${TOUCH_MONITOR}.wav Poblem is of outgoing calls:...

Hi list, The default file name format for touch monitor (automon) recordings is: auto-${EPOCH}-caller-calee It's possible to use the ${TOUCH_MONITOR} variable to change the 'caller-calee' part, but what about the 'auto-${EPOCH}-' part? I've been trying to use ${MONITOR_EXEC_ARGS} to add some more commands after the somix sequence for mp3 conversion. This should work, but I've so far failed to produce any mp3 files...

...started, but my CDR is now full of "s" as the DST instead of the actual dialed number. As I understand it - it is because it is being recorded in the CDR while in a macro (as below). Is there any work around so that I can record the actual dialed number? [macro-dialout] exten = s,1,Set(TOUCH_MONITOR=${TIMESTAMP}_${CALLERID(num)}-${ARG1}) exten = s,n,NoOp(CID_NAME : ${CID_NAME}) exten = s,n,NoOp(CID_NUMBER: ${CID_NUMBER}) exten = s,n,NoOp(CID_CLIR : ${CID_CLIR}) exten = s,n,NoOp(TRUNK : ${TRUNK}) exten = s,n,Set(CALLERID(name)=${CID_NAME}) exten = s,n,Set(CALLERID(num)=${CID_NUMBER}) exte...

Hi list, The default automon (touch monitor) output file name format is: auto-epoch-caller-callee.wav A variable is available to modify the second half: auto-epoch-${TOUCH_MONITOR}.wav But, I can't modify the first half, 'auto-epoch-', with any variables that I know of, including ${MONITOR_FILENAME}. I want to immediately convert this output file to mp3, e.g. with ${MONITOR_EXEC_ARGS}, but I can't because it's impossible to predict a file name that...

Hello list, Using Asterisk 1.2.29 I use the Monitor() application. In extensions.conf I have set MONITOR_EXEC to my script (for mixing files together and convert to mp3) and I set TOUCH_MONITOR on every new channel which has to be recorded. But sometimes I'm missing the recording files. I had a look to the Asterisk-Log and saw those lines: Feb 10 15:18:57 NOTICE[16772] res_monitor.c: monitor executing /root/recordings "//var/spool/asterisk/monitor/XXXXX_20090210-151258-in.wav&quo...

Hi there, We used to record all the calls with the Monitor function. Now, I haveimplemented on-demand recording with automon instead... Everything is working fine apart from the generated filename, which as per all docs, should be auto-epoch-caller-callee....but in my case, it is auto-epoch-who_started_record-the_other_end. Thank you all in advance. Regards, Marta -------------- next part

...(SIP/31247110460&SIP/9083XXX&SIP/0699291034&SIP/webcalldirectDE&SIP/webcalldirectNL&SIP/iXcall&SIP/messagenet) -- AGI Script Executing Application: (macro) Options: (dialout|247110358) -- Executing Set("Local/247110358 at DefaultOutgoingRule-aa84,2", "TOUCH_MONITOR=20070827-224353_0031247110570-247110358") in new stack -- Executing NoOp("Local/247110358 at DefaultOutgoingRule-aa84,2", "CID_NAME : Katie") in new stack -- Executing NoOp("Local/247110358 at DefaultOutgoingRule-aa84,2", "CID_NUMBER: 0031247110460&q...

...-0000017b;1 : -- Executing [9991416445 at from-internal:1] Macro("Local/9991416445 at from-internal-0000017b;2", "user-callerid,LIMIT,EXTERNAL,") in new stack -- Executing [s at macro-user-callerid:1] Set("Local/9991416445 at from-internal-0000017b;2", "TOUCH_MONITOR=1429224932.21697") in new stack -- Executing [s at macro-user-callerid:2] Set("Local/9991416445 at from-internal-0000017b;2", "AMPUSER=9991416445") in new stack -- Executing [s at macro-user-callerid:3] GotoIf("Local/9991416445 at from-internal-0000017b;2&quo...

...rnal-original:2] Wait("SIP/41712-000009e4", "1") in new stack -- Executing [*78 at from-internal-original:3] Macro("SIP/41712-000009e4", "user-callerid,") in new stack -- Executing [s at macro-user-callerid:1] Set("SIP/41712-000009e4", "TOUCH_MONITOR=1373990250.7222") in new stack -- Executing [s at macro-user-callerid:2] Set("SIP/41712-000009e4", "AMPUSER=41712") in new stack . . . Set("SIP/41712-000009e4", "DB(DND/41712)=YES") in new stack -- Executing [*78 at from-internal-original:5] Se...

hi ppl, i'm using asterisk 1.2 because i'm making use of voiceone, but before i was using asterisk 1.4 and had the same problem, it concerns an italian voip/sip provider called eutelia/skypho, my problem is the following one: when i start my pbx my skypho account is working fine, meaning that e.g. incoming calls are shown in the asterisk CLI and caller and callee can hear each other when

...n => s,25,Noop(SIP_CODEC=${SIP_CODEC}) exten => s,26,Noop(SIPCALLID=${SIPCALLID}) exten => s,27,Noop(SIPUSERAGENT=${SIPUSERAGENT}) exten => s,28,Noop(TIMESTAMP=${TIMESTAMP}) exten => s,29,Noop(TXTCIDNAME=${TXTCIDNAME}) exten => s,30,Noop(UNIQUEID=${UNIQUEID}) exten => s,31,Noop(TOUCH_MONITOR=${TOUCH_MONITOR}) exten => s,32,Noop(MACRO_CONTEXT=${MACRO_CONTEXT}) exten => s,33,Noop(MACRO_EXTEN=${MACRO_EXTEN}) exten => s,34,Noop(MACRO_PRIORITY=${MACRO_PRIORITY}) ; ############################################################################ ; Applications [app] ; ##################...

...w the cli == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Executing [0149xxxxxx at from-internal:1] Macro("SIP/101-00000103", "user-callerid,LIMIT,EXTERNAL,") in new stack -- Executing [s at macro-user-callerid:1] Set("SIP/101-00000103", "TOUCH_MONITOR=1426869820.301") in new stack -- Executing [s at macro-user-callerid:2] Set("SIP/101-00000103", "AMPUSER=101") in new stack -- Executing [s at macro-user-callerid:3] GotoIf("SIP/101-00000103", "0?report") in new stack -- Executing [s at macro...

...asterisk -vvvr Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Executing [0176XXXXXX at from-internal:1] Macro("SIP/300-00000192", "user-callerid,LIMIT,EXTERNAL,") in new stack -- Executing [s at macro-user-callerid:1] Set("SIP/300-00000192", "TOUCH_MONITOR=1427481319.470") in new stack -- Executing [s at macro-user-callerid:2] Set("SIP/300-00000192", "AMPUSER=300") in new stack -- Executing [s at macro-user-callerid:3] GotoIf("SIP/300-00000192", "0?report") in new stack -- Executing [s at macro...

...RTP TOS bits 184 > == Using SIP RTP CoS mark 5 > -- Executing [0149xxxxxx at from-internal:1] Macro("SIP/101-00000103", > "user-callerid,LIMIT,EXTERNAL,") in new stack > -- Executing [s at macro-user-callerid:1] Set("SIP/101-00000103", > "TOUCH_MONITOR=1426869820.301") in new stack > -- Executing [s at macro-user-callerid:2] Set("SIP/101-00000103", > "AMPUSER=101") in new stack > -- Executing [s at macro-user-callerid:3] GotoIf("SIP/101-00000103", > "0?report") in new stack >...

You would need to give more information really. Your sip.conf file listing the entries for the phones especially which codecs are permitted. A copy of the 'asterisk -rvvv' console output when you make the call. On 27/03/15 17:05, Salaheddine Elharit wrote: > please no body has som with aastra can help me in this issue > > 2015-03-26 11:02 GMT+00:00 Salaheddine Elharit >

...g SIP RTP CoS mark 5 >>> -- Executing [0149xxxxxx at from-internal:1] Macro("SIP/101-00000103", >>> "user-callerid,LIMIT,EXTERNAL,") in new stack >>> -- Executing [s at macro-user-callerid:1] Set("SIP/101-00000103", >>> "TOUCH_MONITOR=1426869820.301") in new stack >>> -- Executing [s at macro-user-callerid:2] Set("SIP/101-00000103", >>> "AMPUSER=101") in new stack >>> -- Executing [s at macro-user-callerid:3] GotoIf("SIP/101-00000103", >>> "0?rep...

How many times do you ask for help here before getting a respone? Every single thing I do No matter what I get busy extensions. I am willing to pay someone to help here. Anybody got a clue? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050802/d0d1326c/attachment.htm