similar to: TE411P VPM

Hi - I had a client recently move their asterisk system (asterisk 1.4.26, dahdi 2.2.0.1, aex800 w/vpm module) to a "new" location, a building that's nearly 150 years old. I was not personally able to go there, but the person who did the move said the building's demarc room was "scary"-- water leaks, jumbled and frayed wiring, and all sorts of other fun. The echo on

My TE411p does not seem to like to have two PRIs from different telcos (span 1 and span 2). I can get one working, but not the other. However, if I use vpmsupport=0 when loading the wct4xxp module, they both work. But here is the problem, vpmsupport=0 disables the on board echo cancellation. Any ideas? BTW, here is zaptel.conf span=1,1,0,esf,b8zs span=2,2,0,esf,b8zs bchan=1-23 dchan=24

Dear, I have Asterisk 1.6 with an E1 Digium card with echo cancellation module. So I need to use just the echo cancellation by hardware and disable the echo cancellation by software. I use DAHDI for my telephony hardware. If the lines involved with echo cancel are: In /etc/dahdi/system.conf: echocanceller=mg2,1-15,17-31 In /etc/asterisk/chan_dahdi.conf: echocancel=yes echocancelwhenbridged=no

I am looking at the two Digium cards mentioned and am not sure if the echo cancelation is a must. The sever is a 2.8Ghz P4 with 2 GB of ram, and is only for routing PSTN connections to another * server. I would like the advice of the list before making my final decision. -- Bruce Nortex Networks -------------- next part -------------- An HTML attachment was scrubbed... URL:

I just implemented a system using a TE411P hardware echo cancellation card. Per Digium, I setup zaptel.conf, and zapata.conf the same way as I always have. To my surprise calls out to the PSTN had a terrible echo. 1 - 2 second delay, and quite clear. The echo was so bad that I had to remove the hardware echo cancellation module from the card. We are only using the 1st span of this card right

I too had a server room fry and need to replace h/w. So what specific Dell servers did/do you deploy? Where is the link w/Digium/s Dell caveats? I'm using the Digium TDM400 card w/* > Date: Thu, 13 Apr 2006 19:19:13 -0500 (CDT) > From: Aaron Daniel <amdtech@shsu.edu> > Subject: Re: [Asterisk-Users] Digium cards, so disappointing ! > To: Asterisk Users Mailing List -

Hi All, I have installed asterisk 1.6.2.8 Dahdi: 2.4.0 Digium card: Digium, Inc. Wildcard AEX800 8-port analog card I have configured this card properly and it is working for calls too, But there is issue of one way audio on this outbound routes only, My voice is going to outside pstn number but i am not able to get their audio, I have disabled firewall, selinux is also off. If I am applying

Here's an interesting question: If I transfer a call from Asterisk system to another with IAX, is there any way I can get control back on the original system? Or.. do I lose control, and the dialplan has to continue on the new system? Scenario is we transfer calls to an Asterisk system that handles ACD queues. If the ACD queue times out, we want to send the caller to voicemail on another

Hi, I am about to pull my hair out after trying to get our PRI up and working. We are switching from a Cisco gateway to an Asterisk box which provides the 23 phone lines for our office. So, because the Cisco gateway is working I can assume I have all the settings right (b8zs, esf, dms100, etc) and the PRI is live (because we are switching over). When dialing from PSTN, I get busy signal. When

Does anyone have any information on how to use DUNDi precaching? Mark Spencer made a post 2 years ago where he hinted it may be possible to configure DUNDi such that you could centralise your DUNDi registration info by using precaching, instead of having each DUNDi peer meshed with every other one... http://lists.digium.com/pipermail/dundi/2004-October/000189.html However, it seems that no

I changed from a TE410P to a TE411P and fax carriers weren't detected anymore ! I have tried everything (recompile zaptel+asterisk+spandsp ; echocancel=yes/no ; rxgain=0/txgain=0 ; increase Wait() ; ...), nothing worked. The only solution that worked for me was to install and use NVFaxDetect. HTH.

Hello all, I first want to thank everyone for all your contributions. I've building an asterisk system for a month or so now and without everyone in the online asterisk community I wouldn't have made it this far yet. Thanks! ...ok, mushiness out of the way.. :) I am looking for a failover and ultimately a load balancing asterisk solution. I've done a good bit of research and I

Anyone know if a way to have voicemail files stored as mp3's? Thanks, Doug.

Has anyone successfully gotten two separate servers to handle voicemail on an NFS shared mount? The main thing I'm concerned about at this point is keeping both systems from writing the voicemail file to the same filename... any thoughts? -- Aaron Daniel Computer Systems Technician Sam Houston State University amdtech@shsu.edu (936) 294-4198

Does anyone have the polycom soundpoint ip's successfully remotely provisioning? I've got the phone pulling default configs, and it's downloading phone specific information, but it's not actually using that information. Any help would be appreciated :) -- Aaron Daniel Computer Systems Technician Sam Houston State University amdtech@shsu.edu (936) 294-4198

Hi, 1. Has someone met any success at all, connecting a Digium B410P to a Patton Smartnode 4638 (with latest 5.3 firmware) ? 2. If positive, then, which signalling was used on both sides ? My project's goal is use a Patton Smartnode 4638 to act as telco BRI lines, from a B410P-enabled asterisk box. In my testings, I can see channel is respectively up (with patton web server status page)

Does anyone know if asterisk can detect and handle if a phone is forwarded in the dialplan? Aaron

Arrgh. I just made a call with Asterisk to extension 2944093. That extension exists in astdb and I have rtcachefriends=yes in sip.conf. Asterisk did a database query... SELECT * FROM ast_sip_users WHERE name = '2944093' Uhm... Why? Doug

Ugh. We thought we'd fixed some problems by using regexten and DUNDi. Guess not. We have a configuration with three Asterisk boxes. Phones register with a single, primary asterisk box under normal conditions. For voicemail deposit, retrieval, we trunk the calls over to our asterisk voicemail server. However, the voicemail server now has no knowledge of the location details of the phones,

Figured this was worth passing on... This was reported due to the proxy IP address showing up in CallerID on the phone. -----Original Message----- Sent: Thursday, March 23, 2006 12:01 PM Tim, I have tracked down the source of the change in the SIP firmware. The behavior was changed as a fix to bug id CSCsc22406 (host part of the callerid not preserved in ReceivedCall entry). This was a