similar to: newbie question

Ok, I am looking through the iax.conf file now and see 'guest' with no password and tried to add another but none of these seem to let me log from my softphone. I did restart asterisk each time. Thanks! On 11/28/06, blackwater dev <blackwaterdev@gmail.com> wrote: > > I have asterisk installed and now want to try it out. I installed the > sample files and downloaded

Does anyone know of any good step by step tutorials on getting sip set up? I have asterisk installed but can't seem to figure out how to get an account set up and connect from my xTen phone so I can try the demo. The tutorials I read online seem to go into voicepulse stuff and all and I don't have an account there so am a bit lost. Thanks! -------------- next part -------------- An HTML

I have the following code. When I call the extension, it either ignores the first "Hello there everyone", or says "hello" and moves on sometime stoping before it finishes hello. The rest of the text reads fine. Anyone else have this issue?? Thanks! require('/var/lib/asterisk/agi-bin/phpagi.php'); $agi = new AGI(); $agi->answer();

Hi Vince - > My next goal is to setup 1 SIP channel, and be able to call the > Asterisk PBX > from a softphone. > > Then setup 2 SIP channes and be able to call one from another. > > What is the best open source softphone software available for this? > > And what is the best documentation source for finding out how to setup > the > channesl and Asterisk in

I have this code which was taken from the phpagi project page along with the following in extensions_conf and the output from the asterisk CLI. When I call the 311 extension, I does nothing then hangs up. What am I doing wrong?? ----php code------------ #!/usr/local/bin/php -q <?php set_time_limit(30); require('phpagi.php'); $agi = new AGI(); $agi->answer(); $cid =

(Im' answering cc the list, so the knowledge keeps there, and maybe some more qualified answers become). Am Mittwoch, den 15.10.2008, 18:00 -0700 schrieb Francisco del rosario: > Hey Rodolfo... Need some help from you ... > I need to know what hardware do I need to make SIP calls if I set-up > asterisk > So the situation is that I have a PC and configure the software of my PC to

Hello-- What I'd like to do: Use IAX softphones running on computers, in Auto-answer mode, with sound going to speakers, as a sort of public announcement system. What isn't working: Well, my first experiment was to set up the MeetMe system described on the Wiki... This works fine for voice announcements. You pick up a phone, dial the right extension, and an agi is fired up to put files

Hi, sorry for annoying question; i read http://www.voip-info.org/tiki-index.php?page=PBX%20Call%20Pickup without understanding: 1. how to add an ext. to a pickup group (ie:. how to populate pickup group) 2. how 'Directed pickup' does work? "You dial the pickup number and your extension, and the call will only transfer if it is your extension" should i digit something like

Hi This is hard work :) I have read the Asterisk Handbook, BudgeTone User Manual, Andy Powell's useful notes, Zac Sprackett's Asterisk Resource Pages and more. I am not a linux newbie but am new to Asterisk. I have failed to find any docs that explain how to get a very very simple, minimal, system up and I am trying to get the following to work: 2 BudgePhone 102D connected on a LAN to a

Actually my problem is not related to sip.conf and extensions.conf. I have used only standard files from martin pdf which are given as example. I am able to call some system connected over LAN, when each has a softphone and are registered on a asterisk server. But now what i want is instead of using the softphone I write a function in my file which will be executed when the call is placed. In that

Ok, I have trixbox working how I want. How do I now (cheaply as possibly) get a phone number so people can call it from any number? I am just doing a prototype so just want it done cheaply so I can demo it to my supervisors. Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi - after hearing others rave about * I thought I'd have a go - extract from a 'make' on a stock debian system as follows... (I tried to post the whole make up to this point but it was too big for the list) make[1]: Leaving directory `/usr/src/asterisk/channels' make[1]: Entering directory `/usr/src/asterisk/pbx' gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes

I just got cepstal working fine in the dial plan using code like: exten => 511,5,AGI(cepstral.pl|Welcome to my house finder. At the beep enter your zip code.) The php script it calls is based on the nerdvittles weather one so it calls a webpage which prints to the screen, the nerdvittles code uses system to generate the .wav file then has the dial plan call it via: //php script $retcode2 =

Hi..... I am new to this software, and I want to implement a client (SIP or IAX) with PHP or at least to pass the main functions (connection,call, transfer, hangup, call id etc) to a CRM. Does anyone know if I could achive a project like that with AGI ? Any example using AGI with PHP ? Do I have all the functionality with AGI ? What about call id ? What is depend on ? (As I know * does not

I have call center PCs that switch between an IBEAM SIP softphone and a NEBU IAX softphone (for reasons that aren't germane here). The SIP softphones work fine, but the IAX softphones get a fast busy unless I give them an IAX trunk to use, instead of the PRI trunks that all the other phones are using. I am using Asterisk 1.2.3. svn rev 47264. I've appended a sample call trace. The

I have had some radom occurances of someone calling in, and for whatever reason the person is getting dumped into the section of the lcr.agi file that output the message of "it is necessary to dial x". I was curious, does any one know about a variable that might be available in the AGI that would tell me what channel they are coming in on. I looked in the wiki and found

Hi, I have some problem to get this setup working. I have a CAC Channel Banl I, with FXO and an Asterisk box ( I am using Asterisk@Home 1.2) and I have a TE110p installed in this box. What I want to do is, just to be able to dial one of those lines that already are connected to the channel bank, and transfer that call through TE110p and Asterisk to a user agent somewhere through Internet.

You may loose 'control' of the call but you can always 'get it back' Use the UnigueID of the call to track it throught Asterisk. You can palce a monitor event to redirect, bridge, drop, answer or antything else. > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > Sebastian

Thank You . But now i get solved with that error since I had some mistakes in installing googletts.agi Now when calling from my softphone i have written dialplan with an AGI script to convert from text to speech. It get executed without error but there is no sound getting played. My output, == Using SIP RTP CoS mark 5 -- Executing [1310 at Client-dial-Menu:1]

Hello, To do some testing with Asterisk installed the latest Asterisk @ Home in a Vmware system. All worked fine, I can access the web interface (AMP). I have setup the extention and X-Lite softphone according to the description in the Wike (http://www.voip-info.org/wiki-Asterisk+phone+xten+xlite). I can dial 200 (the softphone extention) and 1234 and they connect (the softphone shows this, as