Re: [asterisk-users] Finding the status of an extension

Hi, Here is a small AGI script that get you the hint status of the extension simply call AGI(script.agi,SIP/100) !/usr/bin/perl # # page.agi - Original file was allpage.agi by Rob Thomas 2005. # With parts of allcall.agi Original file by John Baker # Modified by

[asterisk-users] PRI/T1 data rate...

Hi all, First, this is not my first PRI/T1 Asterisk deployement. Did several with Bell, Telus, AllStream, Rogers but this is my first with Videotron. Just spoke with the person taking the order and on top of the standard settings (switch, coding,...) she asked me about data rate (56k or

Re: [asterisk-users] PRI/T1 data rate...

Thanks to all that responded so quickly. It was helpfull to me and I hope other that will be asked the same question by telcos. Andre Courchesne ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To

[asterisk-users] SMS

Hi, Anyone has details or information on how to use the SMS command to send SMS to Fido, Bell Mobility and Rogers Wireless in Canada? Thanks, Andre Courchesne ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing

[asterisk-users] Queues: Play a list of sound file n round-robin at a specific interval

Hi, Anyone knows if there is a way to play a list of sound file in a round robin mode (at specific interval) while someone in waiting in moh in a queue? Ok, you enter a queue and wait listening to moh, every X minutes a sound file is played from a list of sound files to be played. If

Re: [asterisk-users] Queues: Play a list of sound file n round-robin at a specific interval

. Lenz wrote: Why don't you make up the MOH in order to play your sound files, as you need? l. On Mon, 07 May 2007 16:29:28 +0200, Andre Courchesne - Consultant [EMAIL PROTECTED] wrote: Hi, Anyone knows if there is a way to play a list of sound file in a round robin mode (at specific

[asterisk-users] Channel stuck with call pri flag

Hi, I have a problem where some PRI channels get stuck in a Call mode. If I do a zap show channel XX, it shows as PRI Flags: Call. However there is no calls on that channel. Trying to force a hangup does not work: [EMAIL PROTECTED] Dialer]# asterisk -r -x soft hangup zap/27-1

Re: [asterisk-users] Channel stuck with call pri flag

Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Andre Courchesne - Consultant Sent: May 1, 2007 12:44 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Channel stuck with call pri flag Hi, I have a problem where some PRI channels get stuck

Re: [asterisk-users] Trixbox/FreePBX

Hi, If this is with an analog card, the 2 ring is normal since the callerid/callername is transmitted on the second ring. Andre Courchesne - Consultant http://www.net-forces.com -- Message: 7 Date: Sat, 28 Apr 2007 13:33:21 -0400 From: [EMAIL PROTECTED

[asterisk-users] (OT) Madge LMC 10.0

Hi, Anyone has access to the LMC 10.0 software needed to configure the Madge AccessSwitch 20 ? We bought one from ebay last year and now I can not find the CD with the software... Thanks, Andre ___ --Bandwidth and Colocation provided by

[asterisk-users] Queue log analyser / report

Hi, ANyone knows of a fully GPL/OpenSource queue report/analyser project ? Asterisk-stat uses Postgres and that's a no go for us + it's not open QueueMetrics is not Open at all. Any suggestion welcome. Andre ___ --Bandwidth and Colocation

[asterisk-users] Queues Service Level

There seems to be something about SL for queues since when the show queues CLI command is used, it give something like SL:0.0% within 0s: pbx*CLI show queues 1has 3 calls (max unlimited) in 'rrmemory' strategy (243s holdtime), C:174, A:9, SL:0.0% within 0s Members: SIP/1242

[asterisk-users] Detecting no answers and/or disconnected numbers

Hi, Using call files, is there a way to identify no answered calls from disconnected numbers (no longer in service). Both return the same value and so far I can not find a way to know one from the other. Thank you, Andre Courchesne ___

[asterisk-users] safe_asterisks pawning multiple asterisk process???

We have 1 server that after a few hours operating has multiple process of asterisk running. Here is the pstree output: # pstree init-+-atftpd |-auditd---{auditd} |-bash---safe_opserver---op_server.pl |-crond |-cwASTcall.pl |-dbus-daemon |-events/0

[asterisk-users] Returncode from command

Hi, How can I use a command return code in my dialplan? Example, I want to use the system command to run a perl script. This script exists with a code that I need to use in my dialplan. But I can figure out how to extract this value. Thanks for any pointers. Andre Courchesne

re: [asterisk-users] Live creation of trunk groups

Well it works. If I have group=0 that includes all my channels, I can create group=1 which is a subset and a simple reload makes this g1 available to dial on that subset. Message: 12 Date: Mon, 30 Oct 2006 15:25:06 -0700 From: Alyed Tzompa [EMAIL PROTECTED] Subject: re: [asterisk-users] Live

[asterisk-users] Live creation of trunk groups

Hi, Is there a way to create trunk groups while asterisk is running. For exemple let's say that zapata.conf defines g0 as channels 1-23 I would like (while asterisk is running) define g1 as 1-10 and g1 as 10-23 Any hints appreciated. Andre Courchesne

[asterisk-users] Using variable as a context extension ?

Hi, Is it possible to use a variable as a context extension? For exemple: [some-context] exten = s,1,Background(some_prompt) exten = ${key1},1,Noop(User pressed ${key1}) exten = ${key2},1,Noop(User pressed ${key2}) If now anyone can suggest how I could achieve this? --

Re: [asterisk-users] Using variable as a context extension ?

that sets it or by setting it staticly. - Original Message - From: Andre Courchesne - Consultant [EMAIL PROTECTED] To: Asterisk Users List asterisk-users@lists.digium.com Sent: Sunday, October 22, 2006 2:29 PM Subject: [asterisk-users] Using variable as a context extension ? Hi

[asterisk-users] PRI channel becomming unavailable

Hi, We are writing an broadcaster software and under mid-heavy load (dialing on 3 PRIs) we are seeing channels becomming unavailable one after the other untill no more channels are available to dial on. Any ideas or hints? We are running latesr libpri, zaptel and asterisk. Andre

[asterisk-users] MOH problem

Hi, I have a problem with MOH. It plays nice is over the soundcard speakers (using the same command line as what asterisk spawns). But in the phone it sounds distorted or very slow. Any inputs welcome. Andre Courchesne ___ --Bandwidth and

[asterisk-users] Variable that gives the SIP channel

Hi, I have a dialplan code to flash hook from a SIP phone. Everything works great except that it requires the SIP phone to have 2 lines since when the call comes back after the dialplan flash hook, the 1st line instance on the SIP (softphone) is still active. What I would like to do is in

[asterisk-users] IP interface box for Meridian type digital phone

Hi, Anyone can point me to a product that would allow to connect Meridian type digital phone to an Asterisk PBX. I am looking for something like an ATA that you would connect the digital phone to and the ATA would attached to the IP network going to the Asterisk server. Thanks for any

[asterisk-users] Double dial dtmf sounds

Hi, I have site using only softphones (SJPhone under Windows). Once in a while the users complain that they hear double and triple dial dtmf when they dial out. What could be causing that on the asterisk side? Andre Courchesne ___ --Bandwidth

[asterisk-users] Asterisk time not the same as unit time ?

Ok, I test the Asterisk time using SayUnixTime and see the following: -- Executing SayUnixTime(SIP/1000-0822ec80, ||ABdY 'digits/at' IMp) in new stack -- Playing 'digits/day-1' (language 'fr') -- Playing 'digits/mon-7' (language 'fr') -- Playing 'digits/14' (language 'fr') --

Re: [asterisk-users] Asterisk time not the same as unit time ?

not the same as unit time ? To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1; format=flowed Andre Courchesne - Consultant wrote: [EMAIL PROTECTED] tmp]# date Mon Aug

[asterisk-users] Reason to hit failed extension

Hi, Is there any variable set that would indicate the reason why an call initiated by a call-file hit the failed extension? Thanks, Andre ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or

[asterisk-users] IAX trunk behing NAT with dynamic IP

internet IP address changes. My host line is set to a dynamic DNS entry (with zoneedit.com) How can I resolve this so that serverA see the IP address change of serverB ? Andre Courchesne - Consultant http://www.net-forces.com ___ --Bandwidth

V: [asterisk-users] IAX trunk behing NAT with dynamic IP

Would that show serverA the NAted IP address (192.168.10.xxx) or serverB ? Message: 14 Date: Tue, 8 Aug 2006 14:16:10 +0200 From: Jon Sch?pzinsky [EMAIL PROTECTED] Subject: SV: [asterisk-users] IAX trunk behing NAT with dynamic IP To: Asterisk Users Mailing List - Non-Commercial Discussion

[asterisk-users] Agent penality for dynamic agents

Hi, We are using queues with dynamic agents (login and logout using AddQueueMember and RemoveQueueMember). How can setup penalties in such a scenario ? Andre ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list

Re: [Asterisk-Users] Sangoma A104D is dropping DTMF digits, during IVR

Try the relaxdtmf settings. It worked for us. Andre ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Sangoma A200D problem

Well, looks like we had a similar issue. Replaced the Sangoma and it worked. We have asked for a failure analysis from Sangoma on the defective card. Dr. Michael J. Chudobiak wrote: I've been having problems with my A20002D lately - callers from the PSTN don't hear me when I answer, but I hear

[Asterisk-Users] Re: Sangoma A200D problem

Hi, Last time I had this problem was following a unclean powerdown and the solution was: - Kill Asterisk - Stop wanpipe - cd /etc/wanpipe/wan_ec - In there there should be 2 files: wan_ec_pid wan_ec_socket= - Delete those files - Perform a reboot of

[Asterisk-Users] UpState NY SIP provider

Hi, Anyone has good/bad experience with SIP providers in upstate NY? Any recommendations of such provider who works great with Asterisk? Thanks, Andre Courchesne ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing

[Asterisk-Users] Dialing status detection

Hi, Anyone has hints to share about dialing result detection. By that I mean the ability to detect what answered: - Human - Answering machine - Fax - Disconnected number. Any hints or pointers appreciated. Andre Courchesne ___

[Asterisk-Users] Call Transfer Disconnect (CT-5)

Hi, Anyone has experience in using Call Transfer Disconnect (CT-5) over a PRI with Asterisk ? Call Transfer Disconnect allows you to transfer a call to a third party and disconnect yourself from the communication and also freeing your PRI channels. Here is a document that explains how

[Asterisk-Users] Voicemail being cut-off

Hi, I have 2 installs complaining of the same problem. They are both using Asterisk 1.0.10. They complain that when someone leaves a message, they are being cut-off. We tried playing with the maxsilence, silencethreshold and maxmessage without sucess. Any hints? Thanks, Andre

[Asterisk-Users] Ring a grop of extension, then playback a file, then transfer to external number

Ok, Here is what I got working: A call comes in from a Zap line. 5 SIP extension ring if nobody picks up, the call is transfered to a cell phone number. That works. I not want to add a playback of a file (Please waite while you are being transfered) before transfering the call to the

[Asterisk-Users] Display Confideltial or unknown on called id display

Hi, When making a call from an Asterisk box over a PRI connection, I am able to set the Caller ID phone number to what ever I want. This works find. How to I make the called party callerid display Confidential or unknown as we sometimes see ? Andre

[Asterisk-Users] Playback sound file while on-line

Hi, I am looking for a way to play a sound file (wav, gsm or whatever) while a SIP client (extension) is on-line with a Zap channel. Ideally both ends would hear the sound file. Any hints or pointers appreciated. Andre ___ --Bandwidth and

[Asterisk-Users] ZapBarge but ability to talk to the agent

Hi, Is there a way to do a ZapBarge, but where the person doing the barge-in would be able to talk to the agent only (whispering)? Thanks, Andre Courchesne - Consultant http://www.net-forces.com ___ --Bandwidth and Colocation provided

[Asterisk-Users] Dial from php

Hi all, Here is the situation. Linux workstation access a web page on a web server (not the one running Asterisk). From that web page, we need to initiate a dial-out on the Asterisk server and once the call is connected, it must ring on the agent's hard phone. Any pointers about how to

[Asterisk-Users] Play wav while in connection with a caller

keyboard and he hears the sound file and after we can continu talking. Any hints appreciated. Andre Courchesne - Consultant http://www.net-forces.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE

[Asterisk-Users] Zaptel 1.2.3 with Asterisk 1.0.9

Hi, I would like to try the new echo cancelers in zaptel 1.2.3, but don't want to switch to Asterisk 1.2.x just yet. Anyone can tell me if zaptel 1.2.3 will work with Asterisk 1.0.9? Thanks, Andre ___ --Bandwidth and Colocation provided by

[Asterisk-Users] Call Parking...

Call parking... I can park a call that was received on a particular phone. But I can not park a call from the phone that initiated a call. The DTMF are just sent out to audio channel. Any hints anyone? Thanks, Andre ___ --Bandwidth and Colocation

[Asterisk-Users] Up to 4 seconds delay to play prompt?

Hi, Some background... I have the following directories: /var/lib/asterisk/sounds/custom/ - Here are french prompts /var/lib/asterisk/sounds/custom/en - Here are the english prompts If I do: SetLanguage(en) Playback(custom/mypromp) The prompt file is played

[Asterisk-Users] Outgoing fax detection

Hi, Any pointers on implementing outgoing fax detection? Thanks, Andre ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit:

[Asterisk-Users] Write to text file in dialplan

Hi, Anyone has a way to write (append) to text file from the dial plan? Thanks, Andre ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit:

[Asterisk-Users] Voicemail notifications alwats sent as [EMAIL PROTECTED]

Hi, I have a problem with e-mail notifications. For some reason Asterisk does not use the serveremail configuration when sending e-mails notifications. it always send it using [EMAIL PROTECTED] My configuration: pbxskip=yes ; Don't put [PBX]: in the subject line [EMAIL

[Asterisk-Users] Send fax using PRI connection to TE405P

Hi, Anyone has experiences with sending faxes using Asterisk and a TE405P Digium card (or similar PRI) with a PRI connection? Any insights wanted, bood, bad and ugly. Thanks, Andre ___ --Bandwidth and Colocation sponsored by Easynews.com --

[Asterisk-Users] NAT setup

Hi all, I am setting up a a proof on concept where a SIP phone sits on the internet and connects to a * behing a NAT. Right now the SIP phone connects to the * box just fine, I can dial and I see the commands being executed on the * box, but I don't have any audio on the SIP phone. Any

Re: [Asterisk-Users] Montreal usergroup

Hi, You might want to join MLUG which has a lot of VOIP users/experts. http://www.mlug.ca Andre Courchesne - Consultant http://www.net-forces.com Home of the RockHopper Firewall/Server Adrien Laurent wrote: Hi Montrealers ! I would like to create a usergroup for Montreal's

[Asterisk-Users] SIP phone status

Hi, Anyone can point me to a way to get the SIP phones status information (off-hook, on-hook,...). Either through Asterisk or directly from the phone (standard API?). I'm working with the Aastra 9133i. Thanks for any pointers. -- Andre Courchesne