similar to: Recording suddenly stopped

Hello, Information: gcc -v: gcc version 4.3.3 (Debian 4.3.3-3) os: Debian/Testing Pulled latest release from asterisk site, compiled, installed it. I have a barebones configuration: $ ls -l asterisk extensions.conf modules.conf sip.conf users.conf voicemail.conf You can see them here: http://home.comcast.net/~jpiszcz/20090418/extensions.conf

Good evening all, hope everyone in the US had a nice Thanksgiving! On one of our internal servers, I decided to make the leap from 1.4.2x to 1.6.2.0-rc6 so I could start learning about the changes and new features that have been implemented. I upgraded all the configs, removed all the deprecated stuff, etc -- well went well. However, I noticed after the upgrade, when dialing into an

Hi.. I am trying to do something but it is giving me some hard time here. I have an IAX2 trunk to FWD which is registered and working just fine. I have => 011|. as my dial pattern to allow that. But if I want to dial a toll free number I would have to dial 011*1800XXXXXXX What trunk dial rule should I use to enable anyone to call a toll free number by simply dialing 1800XXXXXX instead of

Hi to all, I already setup asterisk on REDhat 9.0 linux machine. I will have 4 physical phone lines and 10 IP phones for it to use. I have a network setup already. Is getting TDM400P - 4port FXO from digium enough to start? Do I need anything else? Thank you

Hello, I have Cisco CallManager 3.3.4 and Asterisk@Home latest version. I have earlier tried getting Asterisk to register with CCM via H323 and failed. Back then, I learned that this is a known bug in Asterisk. Also people who tried doing that had also succeeded in getting calls to go through only one direction like from CCM to Asterisk. I am not that expert so excuse my ignorance with this

Hi, I'm using Asterisk 1.2.1 and I noticed that the voicemail suddenly exits if I press ANY key on the phone while the first or the second voice messages (es: vm-no.gsm or vm-youhave.gsm ) are played. I googled around but found nothing. How can I solve this problem? TIA Giorgio Incantalupo

Hello Everyone, I want to start using Asterisk with Cisco IP Phones 7960 / 7940/ and 7905. Any info or help is appreciated. I know I'll have to use SIP but if I want to use the phones off site meaning from my home for example, how can this be done? Also, regarding external lines what are the options for Asterisk? Thanks Walid -------------- next part -------------- An HTML attachment

Dear list, I am running Asterisk CVS-v1-0-12 what is this called in terms of Asterisk versions convention? Is it Stable , Head, latest release !!! Excuse me if the question is too basic, but your help is appreciated. Thanks Walid -------------- next part -------------- An HTML attachment was scrubbed... URL:

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Hi, Just want to understand the difference between Asterisk Versions and please correct me if I am wrong, I understand they are: Stable CVS CVS Head I am a newbie and about to install Asterisk on SUSE Server. Can someone please advise what is the best version type and number should I use. My environment is not so big. I only wish to eventually get my asterisk to talk to Cisco CCM 3.3.4.

Hi, I am upgrading a Cisco 7960 phone from SIP V.5.1 to 6.0 and then will to go up to 7.5 However in my first attempt to go from V.5.1 to 6.0 this is hat happens: - The phone reboots - The phone then reads the file OS79XX.TXT from the TFP server. In the file I added the version "P0S3-06-0-00" - It starts upgrading firmware - Then I get the following message: (Upgrade Failed -

Hi, I am working on a H.323 project which involves GnuGk and Asterisk My current goal is to provide IVR functionality for the H.323 users which register through GnuGk(eg. call credit information) I have successfully built a H.323 platform using GnuGk - it uses SQL accounting and authorisation. Now I am trying to integrate it with Asterisk in order to provide IVR functionality as I already

Hi all, Any one tried installing Asterisk on Linksys WRT54G? We have but facing problems with SIP to SIP calls. The phones ring and calls are established but we cannot hear any voice at all. I tried allow=all in the general section but did not work. So I forced ulaw. Can any one please check it out and let me know what is wrong? Here are the conf files: Asterisk Version: Asterisk

Hi, I need to setup a demo for asterisk and need some help here please. The demo is connecting to Asterisk a Cisco 7970 SIP (ver. &.0) and a SIP client on HP iPAQ via a wireless hotspot. I need to configure both with the same extension with a shared line like in Cisco CallManager. This way if the extension is called both iPAQ and the IP phone ring and the user gets to pick up using either.

Which version? Similar issues parsing callback number in 1.2.12 > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of Kristian Kielhofner > Sent: Thursday, September 28, 2006 10:27 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [asterisk-users] Voicemail callback

Hi, I appreciate it if someone knows what is available for SIP web phones out there. I am interested in putting a soft phone on a website that registers with Asterisk using SIP. Then, when someone uses it, it directly calls into an asterisk call queue.. Any ideas? -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have the original version of the IAXY. I had it laying around collecting dust, now Im actually putting it to use. When I call my voicemail extension (8500), Before I get the voice prompts from the voicemail app, I hear tones that sound like the caller id tones that are heard when montoring a phone call. While watching my Asterisk CLI, I see this error at the sound of each tone: Jul 21 23:06:03

Hi all. Just to update list and increase the "souls-save" database. The patch solved the problem. Now I have an asterisk-1.2.0-beta1 with asterisk-perl-0.08 and mysql-server-3.23.58-16.FC2.1 machine working fine with ASTCC and "inuse" flag. The link of the patch is: http://bugs.digium.com/view.php?id=5400 Best regards to all you in the list. Ricardo Poppi.

Hi, I'm trying to automatically have the dialplan assign an extension to a roaming phone on my network. I tried the following without success: exten => 3001,1(readop),BackGround(beep) exten => 3001,n,Read(digito,vm-youhave,3) exten => 3001,n,SayDigits(${digito}) exten => 3001,n,Set(ROAM=${digito}) exten => 3001,n,Set(DB(roam/ext)=${digito}) exten =>

Dear sir, after 3 month with asterisk i start SIP A-Z provider for test my solution. Please take a look and test Or send live traffic he he http://msarn.com Dome C. ----------------------------------------------------------------