search for: jonforrest

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

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

...7.0.0/16 da_public group MyROGroup any local group MyROGroup any mynetwork group MyROGroup any dmz view all included .1 access MyROGroup "" any noauth 0 all none none and here is res_snmp.conf [general] subagent = yes enabled = yes Thanks all.! -- *** Forrest Beck IAXTEL: 17002871718 jonforrest.beck@gmail.com

...pt finishes? Then I thought maybe a Macro in the dialplan to dial a global var of the group of phones, but that won't work. If phone isn't available, none will get paged. Has anyone done this before? I just don't know where to start. Thanks -- *** Forrest Beck IAXTEL: 17002871718 jonforrest.beck@gmail.com

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

...e the best way to set the DID for when a extension dials out on the PRI? In sip.conf I am using CallerID as their internal number. I thought of maybe adding a key for each extension to the astdb and have a Macro query the astdb. Any other ideas? Thanks. -- *** Forrest Beck IAXTEL: 17002871718 jonforrest.beck@gmail.com

...f 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: 17002871718 jonforrest.beck@gmail.com

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

...n't need the zttransode module since I don't have a codec translation card. right? To work around this I added zttranscode to RMODULES in the zaptel init script. If I don't need the zttranscode module. I may try and rebuild zaptel without it. -- *** Forrest Beck IAXTEL: 17002871718 jonforrest.beck@gmail.com

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

...s it. After a week or so Asterisk will stop setting the variable MEETME_RECORDINGFILE and start placing the recordings in the sounds directory named meetme-conf-rec.######.wav. Which is the default is MEETME_RECORDINGFILE is not set. Anyone seen this issue before? Thanks! Forrest Beck jonforrest.beck at gmail.com www.shift8.biz #!/bin/bash #Set some variables USFACULTY="ast-phonepages at somedomain.com" LSFACULTY="ast-phonepages at somedomain.com" USFACULTY="ast-phonepages at somedomain.com" MONTH=`date +%B` DAY=`date +%d` YEAR=`date +%Y` HOUR=`date +%I` MIN...

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 --------------

...polycom phones will register with the gateway if the primary server isn't available. They won't have all the features and voicemail, but at least they can dial out and get 911 if needed. What do you think? Do you have a better solution? Thanks!! -- *** Forrest Beck IAXTEL: 17002871718 jonforrest.beck@gmail.com

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

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

...r going to a Norstar Meridian system. The Norstar Meridian is sending a BTN number that needs to be passed to the Telco. Is there a way to pass the BTN as a variable in the dial plan? Like CallerID(num)? What is the variable for BTN if so? Many Thanks. -- *** Forrest Beck IAXTEL: 17002871718 jonforrest.beck@gmail.com

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

...EL 4.0. I will use two "gateways" for my PRI's and FXS Cards so PCI won't be used. I will probably use a small 14" 2U server to handle the ZAP Cards. Does anyone for see a problem with using the 1950? Good/Bad thoughts??? Thanks! -- *** Forrest Beck IAXTEL: 17002871718 jonforrest.beck@gmail.com

...ng else as no answer exten => a,1,VoicemailMain(${ARG1}) ; If they press *, send the user into VoicemailMain [phones] exten => _2XXX,1,Macro(stdexten,${EXTEN},SIP/${EXTEN}) DID example: 2001 = 5552871701 2002 = 5552871702 2003 = 5552871703 Thanks! -- *** Forrest Beck IAXTEL: 17002871718 jonforrest.beck@gmail.com

...cks to this? One I can think of is I will have to specify a extension to redirect 0 (for receptionist) back to the Site A server. I will also have to redirect all directory apps to the voicemail server. Does anyone do this? How do you handle it? Thanks. -- *** Forrest Beck IAXTEL: 17002871718 jonforrest.beck@gmail.com