similar to: automonitor and CDR(userfiled)

Dear, I added new field to cdr table , named "service" and type varchar(20), but in extensions.conf with the following command, nothing to be saved. exten => _X.,1,Set(CDR(service)=OUT) does asterisk support this ability ? is any setting must be changed, before that ? best Mani

hi everybody, I try to record a call with *1 - one touch record feature in g723 format. exten => _00[1-9].,1,Set(TOUCH_MONITOR_FORMAT=g723) exten => _00[1-9].,n,Dial(SIP/${EXTEN}@ext-sip-account,,wW) I have chosen g723 format because in my CLI> show translation there is no translation between g723 format and wav (default for *1 feature). After pressing *1 sequence I have two

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/asterisk-users/attachments/20050219/29dac9ed/attachment.htm

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

Hi, I write the CDR of my Asterisk 1.2.17 server in MySQL database using cdr_addon_mysql.so. Now I'm trying to write some custom values to userfield column by the SET(CDR(USERFILED)=SOME_TEXT) sintax, but nothing gets writeen in MySQL cdr table!! Why? I'm I skeeping something or what? Taking a look at the URL:

Hi, I am using Asterisk 1.8 and trying to pack some custom data in a CEL HANGUP event. In a master (inbound) channel I can set the CHANENL(userfield) to pass custom information to a CEL event. In the outbound channel created by Dial() I can also possibly use a macro/gosub on answer and set the CHANENL(userfield) from there. The problem is how to set it in an outbound channel created by Queue()

I am using VoiceOne http://voiceone.it/ as my management interface. I am not 100% sure when it 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 =

Hi, Asterisk version : 1.2.7.1 stable version We try agent.conf setting of createlink=yes We always can not see this link value to be filled in MySQL's table filed : userfield But we can see the record file has been created correctly. In debug mode, no userfiled shown in SQL command, May 17 18:10:51 DEBUG[2889] pbx.c: Function result is '"unknown" <2001>' May

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

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

hi ppl :D my configuration is as follows, i run (let's call it machine 2) debian etch 4.0 and asterisk 1.2, i use voiceone (www.voiceone.it) as an interface to manage asterisk, I have a grandstream/handytone 486 as a sip device, no PSTN line or anything like that all SIP only. I have a machine (machine 1), which functions as my router and machine 2 and sip device are behind it, grandstream box

I am using Asterisk 11.17.1 with my program that uses AMI Originate calls to generate a bunch of calls for a callcenter. The PBX configuration is handled by FreePBX 2.11. I want to understand the dialplan behavior in order to figure out why the CDR(recordingfile) is blank on the CDR records despite the dialplan setting it. My program generates the calls by setting Channel=Local/NUMBERTODIAL at

Arch x86_64 OS CentOS-6.4 (freepbx) Asterisk 11.4 FreePBX 2.11.0.4 Snom870 with FW-8.7.4.8 What I am attempting to do is to set a different background colour for the BLF vkeys when a station is set to DND. This is supposedly accomplished through this setting in the phones provisioning file: <vkey_blue perm="RW"> DND Blue.on Blue.pickup Blue.park Blue.message </vkey_blue>

Hi, I wanted to revive this issue of supporting asm goto (Bug 9295<https://bugs.llvm.org/show_bug.cgi?id=9295>). As was already proposed, the best way seems to be introducing new IR. If we're changing the IR, we should probably provide an infrastructure that solves or at least enables future support for things like: 1. MS-style inline asm jmps and goto (Bug

thank you for your response below the 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 --

I am making some assumptions, but assuming the 217.195.xx.xxx is your provider, you are getting this back from them: "Got SIP response 556 "No address found" back from 217.195.xx.xxx:5060" Are you sure that "0033149xxxxxx" is the format the provider is expecting? You might try enabling SIP debug on the 217.195.xx.xx IP and seeing what the INVITE looks like, but

I am at wits end...So far it seems dovecot is light years ahead of UW-IMAP. Glad to have UW-IMAP behind me...well almost. My setup = Sendmail 8.13.4 with Procmail delivery to /var/spool/mail/<userfile> openssl 0.9.8 dovecot 0.99.11 I want to continue to use the standard mbox with all the user's mail stored in /var/spool/mail/<userfile>.

Hello Guys, I'm somewhat of a newbie and am desperately seeking for some help... I've managed to get asterisk up and running on my server, and signed up with a broadvoice account... I'm having no problem dialing and communicating between extensions, but whenever anyone tries to call my broadvoice account, they are greeted by no ring or anything, but rather simply a direct to

Linux kernel is using the “asm goto” feature, other projects probably use it as well. I think it provides motivation to support it in LLVM. Regarding the complexity, I believe there is some infrastructure that we can at least partially reuse (the support for “indirectbr” instruction). My focus is adding “asm goto” support, the other things are indeed completely orthogonal and came up in bugs

Hey guys, I''m getting a weird error when trying to test something like the following. It''s a model that is using acts_as_attachment but the weird thing is the equivalent Test::Unit case passes. Does anyone have any ideas? Best, Dave # it was a Test::Unit case first def test_should_be_invalid_without_file UserFile.any_instance.expects(:with_image).never