similar to: Another Asterisk Certification

I was ZapBarging an extension the other day (sorry, CVS Head as of last week) and "forgot" to put down the phone when the conversation ended. To my consternation, I then started to listen to a totally different conversation - I think that the zap channel got re-used. Is there any way to force the zap barge to drop when either party hangs up ? Thanks. Julian

Any means of killing a .call file that is in progress? James Taylor MetroTel 3505 Summerhill Road Suite 11 Texarkana, Tx 75503 903-793-1956

I've got echo problems. *** I'm looking for paid support. *** I'll accept free support, but don't mind paying if someone really knows what they are doing. I've read the wiki, etc. Played with the settings in zapata.conf Using V400P PSTN->_T1->_ASTERISK->_BROADVOICE->_PSTN ECHO ON CALLED PHONE PSTN->_T1->_ASTERISK->_T1->_PSTN NO ECHO

Hi- Can someone help me conceptualize how I could setup an * box which features both English and Spanish prompts? I know it's possible to configure the box to use one or the other, but how could I setup a multi-lingual box? Any suggestions? Thanks in advance, Mike -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello list, Is it possible to implement an application that satisfies the following scenario using agi and php? - user picks up phone - he wants to redirect all his calls to the cellphone - he dials *400 for example and all the calls addressed to him are diverted - he comes back to office next morning, dials *500 and his redirection is gone Note: my extensions.conf is entirely rebuilt at 5

I feel this is a slap in the face for those of us that have been here and I don't feel I should HAVE to pay to be certified... I think me and MANY others are about to walk out of the project over this. I have already spoken with many people that are close to the project. You're hurting US and our ability to make money. I still know the code better than most of the people that will be

I've seen the embedded posts. Is anyone running Asterisk on the MINI ITX? James Taylor MetroTel 3505 Summerhill Road Suite 11 Texarkana, Tx 75503 903-793-1956

Robert, > Is there a product to simulate a PRI trunk? (Robert > Goodyear) TDMoE emulates a T1. ;) Once the TDMoE link is up, Asterisk just sees 24-lines that appear to be a T1 instead of having to deal with all of the complexities of VoIP. This is useful, since probably 75% of the utility of VoIP is really just the fact that it can run over a network. It's also handy because it

Anyone done a chat line app? -- James Taylor MetroTel 3505 Summerihll Road Suite 11 Texarkana, Texas 75503 903-793-1956

I configured this once now I forgot what I did. Two Broadvoice accounts. Incoming is simple - just use the phone numbers. Outgoing: Dial out on a specific line and/or set up the groups and select the other "line" if the first one is busy? -- James Taylor MetroTel 3505 Summerihll Road Suite 11 Texarkana, Texas 75503 903-793-1956

I have added the three lines to the sip.conf file based on the latest changes from broadvoice. I can receive incoming calls but cannot place any outgoing calls. The error I get is: *CLI> -- Registered to '69.73.19.178', who sees us as IPADDRESS:4569 -- Attempting call on SIP/Broadvoice/5068012 for application Playback(demo-congrats) (Retry 1) Mar 8 08:35:21 NOTICE[29290]:

I have a Grandstream GSX-2000 with .. Software Version: Program-- 1.0.0.3 Bootloader-- 1.0.0.3 I tried to do an HTTP update from the Grand Stream web site... After half an hour, I recycled power and now its dead... LED's come on and stay on, screen and buttons are dead. Connectivity to Grandstream.com was always good - whenever I checked (I downloaded the "User Manual" in a

Hi, I have some trouble with the Monitor() application. I start and stop it via the management interface, giving no special parameters except the channel name. What happens is: - if I specify WAV as the format, the resulting files are exactly 44 bytes big and contain nothing at all - if I specify GSM as the format, the resulting files are of size 0. I did not request mixing of the files or

I set up a monitoring system that calls my techs when a problem occurs on one of our networks, everything works fine unless asterisk calls a cell phone in which case the tech can not respond using dtmf. It works fine if the tech call in but not if asterisk call a tech's cell phone. Anyone one have any suggestions? Thanks John Fullington

I'm still having problems. Festival works from command line and I can make the speakers talk. But when I dial my weather extension: -- Executing Answer("SIP/3000-a844", "") in new stack -- Executing AGI("SIP/3000-a844", "weather.agi") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/weather.agi -- AGI Script weather.agi

Has anyone here ever tried using a wifi bridge to place a deskset in someplace where there was no LAN drop? If so what hardware did you use and was it succesful? Michael -- Michael Graves mgraves@pixelpower.com Sr. Product Specialist www.pixelpower.com Pixel Power Inc. mgraves@mstvp.com o713-861-4005

I feel this is a slap in the face for those of us that have been here and I don't feel I should HAVE to pay to be certified... I think me and MANY others are about to walk out of the project over this. I have already spoken with many people that are close to the project. You're hurting US and our ability to make money. I still know the code better than most of the people that will be

i have a model called certification_types that i''ve declared as acts_as_tree. I have a model called certification that belongs_to :certification_type . class CertificationController < ApplicationController ..... def summary_list @certifications = Certification.find(:all, :order=>''date'') end end summary_list.rhtml <% for certification in

Egad, not again with Broadvoice! Anyhow, I recently installed AAH and configured my TDM11B and got that and some SIP phones working. I still have some issues to work out, etc, but my current problem is Broadvoice. I have checked out all of the online resources, including the recent list exchange about the recent changes made by Broadvoice. However, the one thing I have found to be consitent in

Hello! I am in the middle of having a vanity toll-free DID set up. It's been 13 days now (9 business days). This is the first time I'm doing this, and I'm not sure of the process. There has been a very weird progression of changes on my number, from fast-busy, to a message saying that I'm calling from a phone with restrictions (no matter *what* line I call from), to a