similar to: Fw: linksys rt31p2 test case

Hi all, I have installed a TDM22B on an IBM xSeries 220 with FedoraCore 3 and setup asterisk cvs head to work properly. A few days ago, I tried to update the system kernel to 2.6.12 from 2.6.9 and also to change to asterisk stable 1.0.9. After compiling zaptel and asterisk, i loaded zaptel and tried to load the wctdm module but this failed with the following message: Notice: Configuration file

Hi, anyone can confirm if the Linksys's ATA and Router (PAP2-NA and RT31P2-NA) have the same limitation of just one G.729 call like the Cisco ATA 186 ? I'm testing both appliances here and found this issue but could not confirm this anywhere (nothing on the manual, no document or post from any user about this). In my tests they use G.729 only on the first call and G.711 on the

Has anyone tried out the Linksys RT31P2 with Asterisk? Seems like a really great solution for remote users... even supports QoS. Too bad it doesn't also have VPN functionality built in. Here's a link to the product: http://www.linksys.com/products/product.asp?prid=652&scid=29 -Ron -------------- next part -------------- An HTML attachment was scrubbed... URL:

I'm using Asterisk 1.2.3 and PANASONIC KX-TS208W - Speakerphone does not work with it. It works fine when you pick up the handset. Anyone experinced this problem before, the speaker works fine with Verizon line. The phone is behind a Linsys router RT31P2. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hey, I'w got a problem (bug maybe?). I have recently got my Asterisk to work perfect and I'm not trying to setup some dial routes and get the system working as I wan't it to. Yesterday I was installing Festival and also did a "aptitude upgrade" on my Debian Unstable installation. After that the problem started. After some serious testing yesterday night and today I have

Voxilla news story: http://voxilla.com/voxstory84-nested-order0-threshold0.html Two new products * A Sipura 2000 in a linksys box: Linksys PAP2 Phone Adapter * A combination NAT router with 2 FXS ports: Linksys RT31P2 Broadband Router Jim James H. Thompson jht@lava.net

Is anyone else experiencing problems with Asterisk 1.2, the ivr does not work. I have tried it on Linksys RT31P2 and Grandstream Handytone 496. After a call goes through you're not able to enter any of the prompts on a IVR. and cannot enter pin numbers when using a calling card or anything that requires you to enter into an ivr system. I already set my dtmf mode in asterisk. --------------

Can I install the following oH323 software on Asterisk@home: Version 0.7.3 (latest, Asterisk HEAD/v1-2 compatible, date spec 2005-09-08) Version 0.6.7 (latest, Asterisk v1-0 compatible, date spec 2005-09-08) If so, which of the above do I install and what is the difference between the two. Do I also need to install openh323-Mimas_patch2-src-tar.gz and pwlib-Mimas_patch2-src-tar.gz on

Hi; I use the following asterisk, openh323, pwlib: asterisk = cvs-head-03/09/05 openh323 = 1.13.5 pwlib = 1.6.6 asterisk-oh323= 0.7.1 Asterisk, openh323, pwlib were compiled successfully but when I try to compile Asterisk-Oh323-0.7.1 , I got the following error: chan_oh323.o chan_oh323.c chan_oh323.c:37:34: asterisk/channel_pvt.h: No such file or directory ......... ...........

Hi all, I have a problem with voicemail. My boss has asked me to send via email, the message that a user leaves on the voicemail. This is very easy. :) After, he asked me to check before sending the email, if the receiver's mailbox is full. If the mailbox is full, Asterisk should call the receveir intern (example 2001) and using a Playback tell him that his mailbox is full. How can I do?

Hi asterisk GURUs I have an issue with a call forwarding setup, I have the next zapata conf: -----------------zapata.conf----------------- [channels] context=incoming signalling=fxs_ks usecallerid=yes hidecallerid=no echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 channel=1 -----------------zapata.conf----------------- And in extensions.cof I have a

I read the thread about what IP phone is best for business deployment with great interest. Our need is slightly different however. We are deploying VoiP as a value-add with our high speed internet service and are having trouble finding the right SIP analog terminal adapter. In order to support people's existing phones and wiring we need to use an ATA. 1) The first priority is we want

Hello to all, I have a problem with an asterisk qsig. I have three machines: Nortel CS1000 --- Card Sangoma PRI ---> Asterisk QSIG ---SIP Trunk---> Asterisk I use Snom phones on Asterisk. If I call from Asterisk to Nortel, Nortel reminds me of the name of the person i'm calling and I visualize on the display of Snom phone, but if I call from Nortel to Asterisk, the QSIG does not send

I receieved my first PAP2-NA yesterday from our distributor(Tech Data). It installed pretty easily and has worked great so I went to order some more of these units today. When I logged into Tech Data this morning, the PAP2-NA was now marked as discontinued and no longer available and only the PAP2 version was available which is the Vonage branded version. :( I saw someone on the list say that

Has anyone been able to make an html template for the voicemail emails. We would love to customize them beyond just plain text. We have dome some Google searches and have not been able to come up with much. I believe that Switchvox has customized the voicemail email into html. Has anyone ever tried this? Thanks, /Josh -------------- next part -------------- An HTML attachment

Has anyone written a LogWatch script for Asterisk? I use logwatch for monitor all my critical services and would like to do the same for Asterisk. LogWatch is very popular, so I'm guessing that someone has created one but hasn't had time to post it somewhere... Thanks, OCG -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, I have an IAX connection between two asterisk servers and i'm looking for a way to cut down on the needed bandwidth. Both voice and fax calls pass through the channel so it is currently configured to use g.711. Could it be possible to select the codec based on the call's prefix so that g.711 will be used for fax calls and g.729 for voice? Dionisis -------------- next part

Hi, I'm writing an AGI script to manage outgoing calls. We need to interrogate a database to work out which line a particular user is allowed to use for outgoing calls. However, I cannot find a way for my AGI script to access the SIP username. Does anyone know if this is possible (even if it is just passing a variable from extensions.conf to the script)? Thanks in advance, David

Hi, I have a problem with the queue. My system is 'Asterisk 1.8.20.0 built by mockbuild @ buildvm-24.phx2.fedoraproject.org on a x86_64 running Linux' (core show version) and my OS is Red Hat Enterprise Linux Server release 6.3 (Santiago). I have six queues...into the top five I run the application background with a choice (1-5) then redirect to a queue, but here the caller doesn't

Hi, I am using Samba version 3.051 in an Active Directory setting with Windows 2000 server. Everything is working rather well with regards to file-sharing and authentication. However, the one thing that I noticed that I haven't been able to fix quickly with SWAT is the prevention of browsing the Linux file-system with users such as 'nobody' or 'bin'. For example... I have a