similar to: PHP Manager examples

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

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

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..

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 =>

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

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

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 *

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

Hi. There is not much info on the FWD site about this. What codecs do they use? When I try to connect with X-Lite, it works with GSM. When I try to call out with *, it wants G729. I have disallow=all and allow=gsm in the sip.conf. I end up getting errors: Unable to find a path from G729A to GSM Unable to find a path from GSM to G729A What's up with that? I was able to make a call once

Currently at our office, if I want someone else to pick up a call, I have to transfer the call to them. So I'm looking into call parking, which is ALMOST perfect. The missing piece of the puzzle: I'm extension 203. I want any call I park to get parked at extension 2203. I want a call my boss parks to park at 2205, since he's ext. 205. In other words, I want calls parked FROM

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:

Hi. I just tried updating asterisk and I guess I broke something. Here's the log: Unable to find a path from G729A to SLINR Unable to find a path from ULAW to G729A Any ideas on what I should try? I tried nuking all the zaptel stuff in the system and the source and started over agian. Also nuked the asterisk config files.... I saw this asked once before but there was no reply :-/

--- Jonathan Tew <jonathan@ultracart.com> wrote: We've got the app_queue configured to supposedly allow for a call to be transferred. When the call comes in and an agent answers it (using X-Lite Pro) and then decides to transfer the call (using the SIP phone interface) they get disconnected from their call and after left logged in to the queue system. Obviously we're doing

All, I would like to play an announcement to the user on what external line a call came in, right before this call get bridged to this user. How would I go about implementing this in * ? Regards, Hans -- The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may

Hi List, I had a wonderful meeting with GS's President last week and he is very interested in feedback on what top features, functions, bugs the community would like to see in upcoming firmware. Please keep in mind that adding new features take time to develop, test and such. So please rate your ideas on a scale of 1-10 1 = Nice to have some day 10 = Got to have it right now Things

Hi. I just started trying to play with Sphinx. I followed their site as far as running sphinx-server. It is listening on the default port. I copied sphinx2-simple to another file and changed sphinx2-continuous to sphinx2-server. So, I ran eagi-sphinx-test under asterisk. What exactly is it supposted to do? Here's what I get: debian:~# sphinx2-simple2 sphinx2-simple: Demo CMU Sphinx2

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

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 &

Sorry, this is a re-post. I sent the message from the wrong e-mail address. I was checking up on the cvs changes. Hi. I just tried setting up the send url function of Queue app, but it did not seem to work with gnophone. I must have missed something. It never tries to go to the page. I have this in extensions.conf: exten => 1,3,Queue(support,t,http://www.yahoo.com) It shows in the