similar to: Manager.pm port

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 =>

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

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

I've seen references to this module in the mailing list archives, but it isn't in the 0.4.0 tarball, nor is it in CVS. I can roll my own and was planning to do so anyhow, but that doesn't seem to make a lot of sense if it already exists. Am I not looking somewhere I should be looking? Most of the Google hits just point to the mailing list. -- JustThe.net Internet & Multimedia

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

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 *

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

Hi, I've been playing with Voicemail and Voicemail2 a bit for my users, and there are a few things I'm wondering about: - We can specify parameters to the mailbox (s, b or u) to select which prompts to play. However, if we specify 'b' or 'u' it plays that (customisable) message, but it also plays the voicemail instructions. For the dutch, it is customary that a user

Hi, I came back from a quiet weekend today and found my E400P box to have gone astray. Asterisk is loaded from inittab, and started crashing and reloading a couple of thousands of times, each time notifying my monitoring service :-P I remember there would be issues on old cvs stuff since the crash at digium so I made a clean checkout just now. Here is what happens when I load manually:

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

Hi what can be wrong with * that console does not show any stderr text printed from agi script? I am starting with asterisk -vvvvvvvvvvvvvvvvvvvvvvvvrc VERBOSE command does show text on console but printing of STDERR does not I tried it from Perl and from Java and in both cases almost the same result, except in Java more things do not work. In Java for, for example, SAY DIGITS 123 78# would

toy buy my first cisco 186 but when i read this page http://www.djernes.org/~shawn/ata186.htm i cant find in my dev page some parameters just like " UseSIP " what i need to do to show this parameters Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030626/42e66005/attachment.htm

Hello, * newbie here, I'm designing a setup that is to eventually be used in a production virtual PBX/VoIP service. Customers need to be able to change their setups over the web - I want them to be able to do simple things like setting up call forwarding, as well as more intricate stuff that will require me to re-generate their dialplans. Administration of the service is to be

Hi all, DIAX 0.9.4 is available for download from the same place: http://www.laser.com/dante or http://www.geocities.com/tdanro The new DLL contain the latest updates made by Steve in the iaxclient library. What's new in 0.9.4: - IAX2 support (new DLL); - selectable DSP: Echo cancellation, AGC, Denoise; - plaintext and md5 authentication supported; - the phonebook is now in a separate

Hi, Does anyone know of a Print-to-Fax client that works with asterisk & spandsp? Astfax is a partial solution but that only lets us email the fax in, we'ld like to set it up so the user can hit the print button and send the fax (even if all it does is email - transparently to the user - the fax to astfax). -------------- next part -------------- An HTML attachment was scrubbed...

Hi guys, is there any way to execute a command *after* a caller has hung up the call? Something like exten => s,1,Voicemail exten => s,2,AGI(mwi.agi) I'd like to turn off the MWI on my cellphone (which is done by gammu[1]) Or does anyone know a way to check the state of the MWI from outside, i.e. with a cron-job? I'm turning on the MWI with the email-notify from voicemail, but

iaxclient.sourceforge.net is the home of Steve Kann's crossplatform port of the iax library. iaxComm is a client written in c++ using wxWindows. There is a Win32 binary on the site. I think that it should be compilable on Linux and MacOSX, but can't test it. Feedback is welcome.