search for: courchesne

...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 initiale an Asterisk call from a remove web server? Thanks, Andre Courchesne http://www.net-forces.com

...e person taking the order and on top of the standard settings (switch, coding,...) she asked me about data rate (56k or 64k). Since I have never been asked this question before and can find anything relevant in the configuration files, I would like to have your inputs on this. Thanks, Andre Courchesne

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

...o about playing a wav file while talking to someone over a Zap channel ? Let me explain. I am on line with someone. I want him to hear a WAV (or mp3) sound file. I punch a key on my phone keyboard and he hears the sound file and after we can continu talking. Any hints appreciated. Andre Courchesne - Consultant http://www.net-forces.com

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

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

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 cell phone. How can I do this? Andre

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 that possible and if so how? Thanks for any pointers. Andre

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: [root@neil1 Dialer]# asterisk -r -x "soft hangup zap/27-1" -- Remote UNIX connection zap/27-1 is not a known channel Any ideas?

Hi, Ok, here is the setup. Asterisk conected to a PRI line (23 lines). 3 tool-free phone numbers are routed to this PRI line. Customer wants to have a way to have shown on the receptionist phone that the call comes from which of the 3 tool-free lines. Possibly display on the phone that the call comed from tool-free number 1, 2 ou 3 or even better a name or text id associated with this

Prepend *67 if your carrier allows it Thanks, Steve Totaro http://www.asteriskhelpdesk.com > -----Original Message----- > From: Andre Courchesne - Consultant [mailto:courchea@net-forces.com] > Sent: Thursday, April 13, 2006 12:02 PM > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] Display "Confideltial" or "unknown" on called > iddisplay > > Hi, > > When making a call from an...

...sk -G asterisk -vvvg -c asterisk 13681 0.0 0.0 2176 360 ? S 14:11 0:00 /bin/sh -c ( /usr/sbin/sendmail -t < /tmp/astmail-xboGM2 ; rm -f /tmp/astmail-xboGM2 ) & root 13685 0.0 0.0 1820 584 pts/1 R+ 14:11 0:00 grep --color=auto asterisk Any hints? Andre Courchesne

Hi all! Every time i receive a sip call MOH begin to play and i can?t talk to the caller. My setup is the default. Someone knows what is the problem? thanks Miklos iPFONE Telefonia IP Rua Caio Graco 735 S?o Paulo SP iPBX +55 11 3801-3702 FWD 64662 ICH 31451543 www.ipfone.com.br info@ipfone.com.br -------------- next part -------------- An HTML attachment was scrubbed... URL:

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

Hi Montrealers ! I would like to create a usergroup for Montreal's asterisk users. If you are interested, contact me and we'll schedule a beer/coffee meeting downtown next week. Sincerely, Adrien -- Adrien Laurent - CIO 514-284-2020 ext 202 adrien@modulis.ca www.modulis.ca

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

...-- > From: Rich Adamson [mailto:radamson@routers.com] > Sent: Thursday, April 13, 2006 12:14 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Display "Confideltial" or "unknown" on > calledid display > > Andre Courchesne - Consultant wrote: > > 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 "Co...

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

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 1 has 3 calls (max unlimited) in 'rrmemory' strategy (243s holdtime), C:174, A:9, SL:0.0% within 0s Members: SIP/1242 (dynamic) has taken no calls yet SIP/1251 (dynamic) has taken 4 calls

...sterisk 1.4 (have been on 1.2 for a while) and I have a problem where playback and background are played very very fast. When I say fast is you get a few sounds that's it... Running kernel 2.6.20.4 and latest released asterisk packages (asterisk, libpri, zaptel). Any hints? Andre COurchesne