similar to: What is your Backup Strategy?

I am not sure of the best way to do this, so I thought I would query the list. I have about 100 internal extensions ranging from 2000 - 2100. Each internal extension has a external DID number. For example: 2001 = 5552871620. As you can see the internal externsion and DID don't match in any way. What would be the best way to set the DID for when a extension dials out on the PRI? In

Dear list, I am trying to configure the nagios plugin called check_sip. I just read the README file included with the plugin. I follow the readme steps to configure the plugin, without success. In the nagios web interface I can see (No output!) In the status information column. If I run the chech_sip plugin from a linux console, I get /usr/local/nagios/libexec# ./check_sip -u

Does anyone have some method, or AGI scripts that will automatically call a list of numbers from a database and play a pre-recorded message? For example, you have a database of FirstName LastName PhoneNumber Jon -- *** Forrest Beck IAXTEL: 17002871718 jonforrest.beck@gmail.com

I setup sip realtime. Is it possible to use a type of friend? User and Peer seem to work fine. -- *** Forrest Beck IAXTEL: 17002871718 jonforrest.beck@gmail.com

I followed Marks SNMP howto on Voip Magazine and ran into a small problem... (http://www.voip-magazine.com/content/view/2877/0/1/3/) When asterisk is running as a non-root user (asterisk) SNMP request for for the Asterisk MIB tree return nothing. If I quit asterisk and run it as root, all is fine. Does anyone have a idea what is going on? I have never used agentX, so I am unsure of what it is

First off, A lot of thanks to this list. I have learned ton from reading through the posts this past year. I need some advise. I have two group of phones connected to a single server. Group1= SIP/2503&SIP/2504 Group2=SIP/3501&SIP/3502 I'd like to be able to dial an extension and page a certain group of phones only if ChanIsAvail returns 1. I am not sure how to go about

I am looking to setup paging using the auto answer feature on the Grandstream GXP2000. I am thinking I will follow the method as described here: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+page I will setup the 4th account on the phone to auto answer. Does anyone else have a method that works better? I also looked at the allpage AGI written on Voip-Info. But it seems

I have all my SIP users in a realtime database. I would like to use MySQL command to query the database and use the results from the query to page all the phones found in the query. The results from the MySQL query will be multiple rows of extension: Something like: mysql> Select extension from sip where extension like '6%' 6001 6002 6003 ex.... I need to put all the results into a

We are looking to deploy asterisk at one of our locations that will have about 50 phones. I have been buying different phones to test there quality and feature set. So far we have a Grandstream 2000 Grandstream HandyTone 488 Cisco 7912 Polycom SoundPoint IP And we are looking at getting a Linksys SPA-942 Anyone have a favorite? -------------- next part --------------

I know this doesn't belong on this list but... I am looking to see if anyone is using Polycom and knows of a web based software for creating/managing the cfg files for polycom phones. I see that the AsteriskNow will add provisioning support for Polycom phones. Since it is still in beta, I was just looking to see if there was anything else out there. Thanks! -- *** Forrest Beck IAXTEL:

Does anyone have a working implementation of SIP Presence? I have a new Grandstream GX-2000 phone with the supported hardware and I am not sure how to setup presence with asterisk. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060531/4fc3344d/attachment.htm

I was just looking to see if anyone else has seen this problem as well. When asterisk starts up it loads the zttranscode module. The problem exist when I use the init scripts to stop asterisk and then use the zaptel init script to unload modules. Since the zaptel init script didn't load the zttranscode module it will error out when trying to unload the modules. I built

Any idea when 1.4.1 will be available. There is a bug fix in the cvs head that I need, and I don't want to run the cvs build on a production machine. Thanks... -- *** Forrest Beck IAXTEL: 17002871718 jonforrest.beck@gmail.com

Hi all, I am new to Hearbeat so please be kind :) I also posted this on Linux-HA lists with no responses so I posted it here. I have successfully configure two machines to use heartbeat to cluster httpd. The two nodes are called etk-1 and etk-2. I am trying to configure another two machines to act as a separate cluster (on the same IP subnet). These two nodes are called radu-1 and radu-2.

I get the following error when trying to compile zaptel on CentOS 5 kernel 2.6.18-8.1.3.el5 CC [M] /root/asterisk-src/zaptel-1.2.17.1/xpp/xbus-core.o /root/asterisk-src/zaptel-1.2.17.1/xpp/xbus-core.c: In function ? /root/asterisk-src/zaptel-1.2.17.1/xpp/xbus-core.c:171: error: ? has no member named ? make[3]: *** [/root/asterisk-src/zaptel-1.2.17.1/xpp/xbus-core.o] Error 1 make[2]: ***

Is there a way to include a search of a text file in the dial plan? I am trying to think of a good way to keep a sort of Blacklist file that is checked against before letting a call through. If the callerid is listed in the file, it will go to Hangup() -------------- next part -------------- An HTML attachment was scrubbed... URL:

I am having a weird issue with setting the recording file for the Page app. Here is some quick background info I have a macro that pages all my phones: [macro-pageall] ; Context for paging all devices. ; This will search the sip table in the realtime database ; for all phones that start with a number. That number is ; passed to this macro as ${ARG1}. ; ; ARG1 = The

set_format: Unable to find a codec translation path from ulaw to g729 Both endpoints are PAP2 set to G711 only I have 1.2.17 on Suse 10.1

hi! First of all i want to tell i have a dedicated server on layeredtech with direct internet connection and i currently dont use iptables, so this is not about network configuration =). well so, i install asterisk-1.4.2 on my server, and next install asterisk-gui from the digium repository. next i go to: http://pbxa.com:8088/asterisk/static/config/cfgbasic.html and install a default

Can anyone recommend software that will allow me to utilize my VoIP provider and send fax over IP? I use Asterisk now for my phone system. Thanks! Wiley E. Siler Director of Information Technology Education 2020 4110 N. Scottsdale Rd. Ste 110 Scottsdale, Arizona 85251 (480) 296.0260 (866) 320.2083 ext. 1003 mailto:wsiler@education2020.com <mailto:wsiler@education2020.com>