similar to: AMP-1.10.007 Released!

Hello all, Asterisk Management Portal 1.10.009 has now been released. This exciting new version has several notable additions (listed below). The AMP homepage is http://amp.coalescentsystems.ca. Here you'll find links to the download, install guide, and documentation wiki. As usual, please use amportal-users mailing list for discussions about AMP:

I'm looking at the Sayson 480i or the Cisco CP-7960. Which one would you suggest and why please ? Best Regards, Alex Brecher Visit us at http://www.Successfulhosting.com <http://www.successfulhosting.com/> -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041128/8e282c51/attachment.htm

Hi, I have the newbie guide from AMP's website and (fair enough) it is all about whitebox linux. Has anyone found any gotchas with the newbie guide relating to SUSE 9.2 ? Any help appreciated. Cheers -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050125/6b7a2f61/attachment.htm

Does anyone know of any obscur reference for detecting an incoming fax. I currently have AMP running and everything else is working great. Installed the spandsp patches and software... using the default AMP extensions.conf, I start sending a fax, I hear it pick up and transfer to voicemail after 20s. Fax is set for system... Here is the detail from the extensions.conf [global] FAX_RX = system

Hello all, A new version of the Asterisk Management Portal is available for download. Please visit the AMP homepage at http://amp.coalescentsystems.ca Upgrade instructions are at http://amp.coalescentsystems.ca/UPGRADE Use our Sourceforge mailing list and forum for discussions about AMP. 1.10.006 ChangeLog: - Use extensions_custom.conf for customizations. Sample included. - Added option

Hi all, I curious to know if other UIP200 users have this same issue: You flash (XFER button) to consult-transfer a caller to another extension. If the transfer target party is unavailable (ie: voicemail), there appears to be no way to get the original caller back. If it's a known limitation, has anyone come up with a functional work around? Thank -- ..................................

Hello all, The Asterisk Management Portal (AMP) is now known as FreePBX. FreePBX 2.0.1 is now available for download. A **BIG** thank you goes out to the project developers for all their hard work, and to beta testers for running FreePBX through it's paces! This exciting new release boasts a better user experience, additional functionality, and a new module system. The module system is

Hi, all. I've got my Uniden UIP200 configured via TFTP (had to get DHCP 3.0.1 -- Debian's latest is 2.0.x!), and all seems well... except for the minor detail that it doesn't work. It registers fine with Asterisk, but when I copied my Grandstream's sip.conf info and plugged in the Uniden stuff, no dice. Any ideas? Thanks... -Ken unidencom.txt: OverwriteLocalSettings

I'm looking at deploying some Polycom 501's here, but one thing that still needs confirmation before I can move forward is global paging. I figure that I can couple polycom auto-answer (http://www.voip-info.org/tiki-index.php?page=Polycom+auto-answer+config) with this script: http://lists.digium.com/pipermail/asterisk-users/2004-March/040186.html However, that script was posted over a

Hello All, I'm new to the list and the whole voip server side. I'm trying to setup Asterisk to just do internal dialing, no access out to the pstn is required/wanted at the moment. I'm running Fedora Core 3 with Cisco 7960's phones (running SIP 6.3). I've set it up following these guides: http://www.voip-info.org/wiki-Asterisk+Fedora+Core+3

I couldn't be happier with the simplicity of this - but it's a hack! Hi all, I'm currently using a SIP client (BT101) to connect via DSL to a remote instance of Asterisk. - Asterisk has a private IP behind my OFFICE router. - The SIP client has a private IP behind my HOME router. I'm doing this _without_ the use of STUN or proxy servers. Here's how it works: -

Hi all, Several months ago we built an * box with a quad-FXO tdm400p (REV e/f). >From the get-go, there has been a problem where occasionally (2-3 times a week) zaptel/* will not detect the ringing on a line. (The call will ring through to telco voicemail). The problem is not specific to a single line or FXO port on the tdm400p. I have 2 theories: #1 - the ring voltage for some calls is

Hello, Over the past several weeks, we have been having an intermittant problem with our deployment of a TDM400P card (4 fxo). ?We have tried many things, and the problem still re-occurs. The Problem: Occasionally (every 48 hours), the TDM400p card will stop answering incoming calls on ALL fxo ports. ?Attempts to send outbound calls on any Zap channel will result in hearing a loud

Hi, We've recently deployed 6 Uniden UIP200 phones (running firmware 4.54). We've been having some serious problems: 1) All the phones randomly reboot themselves. Typically when trying to answer or initiate a call. 2) All the phones will disconnect from a calls with the PSTN after 2-3 minutes. 3) The phones are unable to interact with a remote IVR (digit presses are not received at

Dear forum, I have a install of asterisk using AMP. I followed the install guide off the AMP site. http://amp.coalescentsystems.ca/docs/AMP_Installation_Guide_v1.4.pdf When I start using amportal start or asterisk -ccccv I received this in my log. The last line is that ogg failed. I have found nothing on the web about this, and I am not even sure where to start troubleshooting. Any help

Hi all, I know that this has been passed around before, and I know that it happens about every 3 months or so, but evertime the answers change, so I thought I would pass it around again. A company I work for has 3 incomming lines and 4 phones. They require voicemail and MOH. Their phone systems VM hard drive died today, they were quoted a $2000 to replace it. I started to talk to them about

Hallo, we have started playing with asterisk about one month ago, and we do like very much what we are experiencing. Now we would like to take some step further towards "standardizing" installed modules, functionalities, tools etc. The "wall" we are facing now is: choosing the right tool for * management. We tried AMP, very powerful but incomplete (CAPI is very important to

I installed oh323 and everything seemed to go smoothly (compile & everything upto calling through using oh323). I must admit, there is some behavior that's doesn't seem right but generally, I'm able to dial-out of any oh323 device whether to an extension or to a trunk. Audio is sometimes muted when dialing out until the extension or dialed number answers. Sound quality is good

I compiled libtiff version 3.6.1 and spandsp and spandsp version k. When trying to load asterisk I get the folloein error: Jun 7 10:15:03 WARNING[16384]: loader.c:408 load_modules: Loading module app_dtmftotext.so failed! Ouch ... error while writing audio data: : Broken pipe [root@zapata root]# Warning, flexible rate not heavily tested! Please help! -- Manuel Marin Garcia TRANSTELCO S.A.

I have Asterisk CVS-HEAD-05/25/04-17:13:22, Copyright (C) 1999-2004 Digium. Usign exclusively digium hardware. 3 TDM400P cards. 1 4xFXO 1 4xFXS 1 1xFX0 & 3xFXS When * is attending FXO calls, bridged to FXS calls, natively ofcourse, at a random time, the call hangus up. Also, for example, if a call is done, and an other extension hangup, there are some probability that the other extension