similar to: Channel stuck with call pri flag

Hi, in a PRI setup, the receiving side is changing the B channel at proceeding. It seems this sometimes breaks some logic (pri_fixup_principle) and then the hangup kind of breaks, release is not answered and a restart cycle is triggered (by remote side). Anyone can help me debug this ? I've seen many posts with simmilar issues but no answer/solution. This is happening on Asterisk 1.2.16 +

Dear All, Is it possible to install * in front of a Avaya IP 406 system via a T connector E1 tap so it's external to the Avaya system? We would like to record upto 60 channels (2 * ISDN30e). This may increase later. Also, could the calls go into the cdr for retrieval/browsing later? What hardware/server would you recommend? Thanks.

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 64k). Since I have never been asked this question before and can find anything relevant in the

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 initiale an Asterisk call from a remove web server? Thanks, Andre Courchesne

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, 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

Hi, For thanks to everyone that answered the "dial from pph". On an other subject, how would I go 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

Just curious: has anyone seen or heard about a USB-based T1/E1 interface device? I've seen some serious T1/E1 testing equipment that is USB-based, but I was wondering if there was something more generic, like a Zaptel-ish T1/E1 that used USB instead of PCI/PCIx. Thanks! -MC -------------- next part -------------- An HTML attachment was scrubbed... URL:

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, Does anyone know of a way to get a dry copper pair (also known as an alarm line) from Verizon for less than $20/end? I know we have been able to get them, but they come out to $40/month for a circuit.. and there's no dial-tone over it!!!! -------------- next part -------------- An HTML attachment was scrubbed... URL:

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 |-hald-+-hald-addon-acpi | `-2*[hald-addon-stor] |-httpd---3*[httpd] |-khelper |-klogd

Hi, Any reason why I can not get the php exec() function to execute a shell command inside an agi script? Thanks. Andre

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

I have a * serving 15 sip clients. I use the digium 4 port t1 card. We have an autodialer that calls and reminds clients of there appointment. it uses a pri t1. I would like to plug its t1 output into asterisk to use voip. I am very new to * and am confused. Any help would be appreciated. _________________________________________________________________ Express yourself instantly with

If you have a PRI terminated in a Cisco router talking SIP to * and would be willing to share your Cisco config, please respond. Also, I would be interested in knowing what version of IOS you are using. We are using an NM-HDV in a 3640. TIA Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815

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

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 has good/bad experience with SIP providers in upstate NY? Any recommendations of such provider who works great with Asterisk? Thanks, Andre Courchesne

I am trying to automatically detect disconnected numbers when using the outbound dialer I have written. * Some numbers hang up immediately with a Cause Code > 0 and no voice treatment * Some numbers get voice treatment with a PROGRESS indication and an associated Cause Code > 0 * Some numbers get voice treatment with a PROGRESS indication and no associated cause code (CC=0) My application

[Apologies if this gets through twice. I sent it first without subscribing, but it seems like it got stuck in the moderation queue, so I subscribed and re-sent it.] I'm doing some work on audio fingerprinting for a school project (more precisely, my master's thesis. I got a hint on #vorbis that I might want to look into the internal floor representations in libvorbisenc to get out audio