similar to: Include Text file in Dial Plan

Displaying 20 results from an estimated 7000 matches similar to: "Include Text file in Dial Plan"

2007 Feb 08
3
Automatic Dial, Play message
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
2007 May 03
3
SIP RealTime Friends
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
2007 Feb 25
1
Marks SNMP HowTo
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
2007 Mar 30
1
Paging
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
2007 Apr 19
3
Outgoing CallerID
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
2006 May 11
2
Paging and Auto Answer on Grandstream GXP2000
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
2006 Jun 28
6
Suggested Phone
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 --------------
2007 Apr 04
1
Polycom
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:
2006 May 31
5
SIP Presence
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
2007 May 03
2
zttranscode crashes server
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
2007 Apr 11
5
What is your Backup Strategy?
I was just curious to what your redundancy solution is. I have considered many options, so I thought I would share and get an idea for what others are doing. My setup is two different locations with a 10MB WLAN fiber link between the two. Each location has it's own PRI as well. I have considered and tested many options this last year or so. 1) Using hearbeat and drbd to monitor the
2007 Sep 20
1
Paging MEETME_RECORDINGFILE Variable
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
2007 May 04
4
zaptel compile error
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]: ***
2006 Oct 31
1
Strange Characters in CLI on TTY9
When I look at TTY9 (using init.d and safe_asterisk to start the asterisk process), I am getting some strange characters. When a application is run the and the CLI shows the application executing the languange almost looks russian...?? Anyone seen this before? http://picasaweb.google.com/jonforrest.beck/AsteriskCLI
2007 Apr 11
10
Nagios asterisk monitoring
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
2007 Jan 11
2
Voicemail IMAP
I know some of this doesn't belong on this list, but I am just including it for problem history. I am trying to setup IMAP Voicemail with our email server. We are using a non-standards based groupware server called FirstClass. The server has some built in support for IMAP. My problem seems to be that the authuser flag is not supported. When I use mtest in the imap toolkit to connect to
2006 Nov 06
1
Register vs. Host=IPADDR
I am not sure if I am going to use SIP registration's or just specify the host ip address in sip.conf. Are there any pros or cons to the two? My phones will have a static IP address and won't be changed unless a admin does it. So the logical path would be to just turn off registration on the sip account (in the phone setup). Can anyone forsee a problem to this? Something I will miss
2007 May 04
2
AsteriskNow!
Does anyone know how to gain access directly to the configuration files in AsteriskNow? I have dual NICs and need to change the binding in the config file. I believe they blocked ssh2 access by default. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070504/c4d288d1/attachment.htm
2007 Aug 15
2
Disable MoH for certain phones
Hi, Is it possible to configure asterisk so it doesn't play MoH from certain phones? Regards, Jan
2007 Oct 06
2
Change verbose level
Hi folks, How I can change default level in asterisk from 3 level to 7level, using the script /etc/init.d/asterisk