similar to: AGI.pm?

and I found a reference to an AgentLogin.rtf. Looks great, except I can't get it to work. queues.conf: [sjs-testq] music = default timeout = 1 retry = 1 maxlen = 0 member => Agent/10001 agents.conf: agent => 10001,1234,Steve Sobol extensions.conf: (I have a phone line set up on which the main menu tells you to press 1 to be added to queue. Pressing 1 lands you here) exten =>

For anyone that cares... I am porting James Golovich's Manager.pm over to PHP. I plan on also doing some documentation which will cover both the Perl and PHP APIs, which will be almost identical (at least, to whatever extent is practical). Will let y'all know when I have some usable code to show you. -- JustThe.net Internet & Multimedia Svcs. [The Fusion of Content &

Can anyone give me a pointer to descriptions of the status codes my Grandstream phone displays? I've looked on Google but can't find a definitive listing of SIP codes. -- JustThe.net Internet & Multimedia Services 22674 Motnocab Road * Apple Valley, CA 92307-1950 Steve Sobol, Proprietor 888.480.4NET (4638) * 248.724.4NET * sjsobol@JustThe.net

Quick question: My PHP script is now able to connect to the manager port and successfully authenticate using MD5. I would strongly prefer not to do plaintext authentication at all. Would anyone object to plaintext authentication being left out? -- JustThe.net Internet & Multimedia Svcs. [The Fusion of Content & Connectivity] 22674 Motnocab Road * Apple Valley, CA 92307-1950 Steve

a call on which caller ID is unavailable, and a call that's supposed to be private? As a side note, I have a phone on which I have caller ID blocked, but the Asterisk server still ends up getting caller ID from that line anyway. -- JustThe.net Internet & Multimedia Services 22674 Motnocab Road * Apple Valley, CA 92307-1950 Steve Sobol, Proprietor 888.480.4NET (4638) * 248.724.4NET *

X-Lite build 1079 consistently chokes no matter which codec I use - after five seconds I suddenly have no sound coming in and possibly no sound going out too. Putting the line I'm on on hold and then switching back to it gives me another five seconds of sound, then it dies, etc. The Grandstream 101 I'm using is a piece of junk but I don't have the same problem with it. Not sure

So I hacked up the Dial app to return a numeric return code instead of changing contexts based on a number being busy or unanswered. The purpose for this modified dial app, which I call AGIDial, is to help me concoct a "follow-me" type of application. The app returns -1 for a completed call, 0 for unanswered, or 1 for busy. Well, I hooked the thing up to an AGI script that uses perl and

Hey all, I am in the High Desert region of southern California, USA. I was wondering if any of the SIP providers offer numbers serviced out of the following Verizon central offices: Apple Valley (Apple Valley CO/APVYCAXF) Apple Valley (Desert Knolls CO/DSKNCAXF) Victorville (VTVLCAXA) Adelanto (ADLNCAXF) Hesperia (HSPRCAXF) These are the COs which offer prefixes which are local calls from my

I can't do threeway from my Grandstream phone. Looking through the server config files, I figured out why - zapata.conf has Threeway turned off for the channels I use. I do my work on someone else's Asterisk box and don't want to modify zapata.conf for several reasons, the biggest being that the guy who owns the box has a couple clients using it and I am deathly afraid of breaking

asterisk-users-request@lists.digium.com wrote: >Send Asterisk-Users mailing list submissions to > asterisk-users@lists.digium.com > >To subscribe or unsubscribe via the World Wide Web, visit > http://lists.digium.com/mailman/listinfo/asterisk-users >or, via email, send a message with subject or body 'help' to > asterisk-users-request@lists.digium.com > >You can

I need to get your thoughts on something.. :) I am trying to create a system to process the CDR call logs for department accounting.. I think there are two ways of doing it.. Either I can create an AGI that will run on the "h" extension and will lookup the last entry that matches the account code of the call that just ended in the MySQL CDR and calculate the call cost immediately..

Anyone have any example scripts in PHP that connect to the manager? I'm not really a much of a programmer so I could use boost. Once I can figure out how to get it to login properly, I'll be ok from there. Thanks, Kevin _____________________________________________________________ Are you a Techie? Get Your Free Tech Email Address Now! Visit http://www.TechEmail.com

Hello Since I am getting a bit concerned about the SCO vs IBM issue, I was wondering if can I can setup Asterisk on FreeBSD is it supported ? Are drivers for Digium cards available on FreeBSD ? Thanks Ajit ----- Original Message ----- From: <asterisk-users-request@lists.digium.com> To: <asterisk-users@lists.digium.com> Sent: Wednesday, July 30, 2003 3:05 PM Subject: Asterisk-Users

Hi, Anyone succeed using call transfer function in X-Lite? It is stated that this feature is available in the Lite version too, but for me it doesn't work. Clicking on Transfer button, then entering the number and then clicking again on transfer doesn't work. I miss something? Thanks, Dan

Does anyone keep a known telemarketer caller id database? If not has anyone proposed an Asterisk community project to share this information? Sort of a nation wide blacklist so Asterisk'ers can cut down on the garbage calls... -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hey all, Did I see something on here about using an AGI script to do reverse lookups via anywho.com? I have a PRI that only gets caller-id number and no Alpha. TIA, -- Daniel Jimenez <djimenez[at]pobox[dot]com>

Hi All, Our FW appears to be blocking my CVS attempts, does anyone have a tar ball they can send me ? Rgds, Adam ********* DISCLAIMER ********* This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message

Hi, I want to unsubscribe from asterisk-users-request-lists, and donot want to recieve mail any more. Kindly unsubscribe me... sanchal singh On Mon, 2006-12-18 at 13:57, asterisk-users-request@lists.digium.com wrote: > Send asterisk-users mailing list submissions to > asterisk-users@lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, visit >

I'm finishing up deploying an Asterisk (Trixbox) box at work. Wow, I thought Asterisk was cool by itself, but Trixbox has made just about everything turnkey. Great stuff! So... we're using Grandstream GXP-2000 handsets to connect to the Trixbox, which sits on our DMZ with a single public IP. I need the phones to work from random places behind NAT, as well as in the office. I'm using