similar to: MeetMe Conferencing - Announce your own join/leave to yourself and other conference members

Greetings, I am writing an AGI script that needs to check on the idle/busy status of a number of SIP peers (mostly SPA9xx phones, with a few Polycoms and Snoms thrown in for fun). Is it possible to call Asterisk functions (e.g. SIPPEER) from AGI scripts? Based on my Googling, I would guess in the negative. I have tried various permutations of Set() and Eval() without success. I have also

First of all, you can probably go online before you convert all passwords. You can modify your query in dovecot-sql.conf.ext to something like the following: SELECT IF(crypt_pass IS NULL OR crypt_pass='', CONCAT('{PLAIN}',plain_pass), crypt_pass) as password FROM mailuser .. This is assuming that: * for incoming users, you have a plain_pass column containing just the plaintext

Hi all, I'm about to start setting up my first asterisk/cti system in our test lab. I've read through all the documentation I can find and relevant posts in the list archives but can't seem to find anything explaining how to go about initiating an http request upon an incoming call. I basically want asterisk to request an uri on our intranet, which will pass call details to our

Here''s the script. I would have posted it to a website, but I figure this''ll be better so it is always on the list for people in the future in case they wanted to see it. If you have any questions about any of it, please let me know. Mike and I aren''t the cleanest of programmers. Cron the script to run whenever you need it to. :) #!/usr/bin/perl # # TC Helper Script:

Hi, Here is my problem. I am trying to get the Status of the call if the user picked up the phone or not. It is coming as empty. Please help. Here is my extensions_additional.conf file code: [multi-dir-callback] include => multi-dir-callback-custom exten => _X.,1,Answer exten => _X.,n,Playback(beep) exten =>

You do need to complete the query. Don't just replace your query with the one I wrote. You have to have a WHERE clause, and you might need to return other fields. Keep the password query you had before, just replace the 'password' column with "IF( ... ) as password" The query as you have it now simply returns all the passwords for all the users, because you don't have a

Here is some code to do authentication by caller ID for DISA through AGI. My original code had a bug in the Mysql query code, and there was a hangup in the wrong place [that's what I get for coding something at 2:00am], but the attached code works correctly. Take note of the REGEXP for the CallerID variable. When I tested the code from the PSTN it worked because there was no name component,

This looks good, except it is truncated, it should be something like 95chars long, Is your hash column set to 128 or up around there or larger? Quoting Carl A Jeptha <cajeptha at gmail.com>: > Sorry for double reply, but this what a password looks like in the > "hashed" password column: > {SHA512-CRYPT}$6$wEn1UFuiMzl9OSjd$Vh/PZ95WDID1GwI2 > > ------------

hi, i use $AGI->wait_for_digit($timeout) to wait for the user press key 1 ,and then to do something. but how can i get the return number ? is that use $key = $AGI->wait_for_digit($timeout) and $key will be "200 result=49" if i pressed number 1? Thanks! -- Best regards, Sucan

Hello everybody!!! I really need your help guys, I am using the AGI mode in meetme application, and I want that AGI should wait for an input from the client/user i.e. a digit and then proceed, but I have used that AGI function agi->wait_for_digit(), but no use....my agi just passes, or ignores this function, where AGI should stop here and wait for the input.... .....my extension in my

Hi there, I am new to this ML. Recently I started working on Asterisk 1.4 + RAGI + Ruby on Rails to create a call history browser. To record call history, I am trying to capture dialup, answer and hangup events. To check what status a call is, I use channel_status() that RAGI provides. I am having a trouble on this function. In a polling loop that checks call status, the first call of

Hello, I am running asterisk 1.4. For argument's sake I have 4 telephones. 2 support video, 2 do not. Calls between phones work fine and codecs are properly negociated. I have videosupport=yes in sip.conf and when the two video phones communicate I have video. I call meet me with this command EXEC MEETME 1234|d SIP looks like this : -- AGI Script Executing Application: (MeetMe)

Hello, Situation: I've got two asterisk 1.2.4 servers, connected to each other over the internet with IAX2 with about 20msec delay. One of the servers is hosting MeetMe. It's working fine as long as only SIP phones connected to the meetme server participate in the conference. As soon as a participant using IAX2 is connecting, lots and lots of buffer overruns and underruns are

I'm trying to fix a problem with the GrandStream Budgetone 102. I've been reading the source code, mailing lists and other resources. Here's the scenario and the approach I have been pursuing. I'm having some problems with the AGI calls and I hope someone can give me some clarification. PSTN <---> T1,PRI * <---> Grandstream BT 102 (12)

Hi, We are trying to implement a complex business logic in Asterisk. Executing "Wait_For_Digit" command after playing IVR. We want to stop the IVR once we receive the digit. It is not recognizing the Digit until it completes the IVR. How can we stop the IVR once we receive the digit? Thanks BB -------------- next part -------------- An HTML attachment was scrubbed... URL:

Good day! Have a weird problem with HT-286 and Conference room. I use Asterisk CVS-HEAD-06/04/04. Here it is: When HT-286 get into the conference room first and nobody in that room everything seems ok (with any codec HT286 allowed), but when HT-286 get into conference room when somebody already there, have got different HT behavior: 1. When HT use GSM codec => it connects to conference room,

I am using asterisk Version: 1:1.4.11~dfsg-1 as found in Debian. I'm using a call file to connect a meetme conference to an AGI script which plays files using the stream_file method. I have four files which should play in sequence, though only the first two files actually play. I get these errors in the CLI: [Sep 16 06:20:43] NOTICE[18424]: app_meetme.c:1911 conf_run: Audio bytes: 276 Buffer

Hello steve. Here is a patch I wrote for app_voicemail.c which does exactly as you describe. When the outgoing message is playing, if the listener hits the "*" key, they're prompted for a mailbox and password, whereupon they can check their voicemail as if they were using the internal phone. I found no other way of doing this. If you patch your app_voicemail.c, I have V1.44 from

Hello! I'm trying to setup a perl-deadagi, but my perl skills lack. can someone tell me why the following code doesn't work: #!/usr/bin/perl use Asterisk::AGI; $AGI = new Asterisk::AGI; $dialstring = $AGI->get_variable("DIALSTRING"); $res = $AGI->exec("DIAL $dialstring"); the asterisk output says: AGI Rx << GET VARIABLE DIALSTRING AGI Tx >> 200

Hi i want use the API on my asterisk 1.6, but i have a small problems : In extension, i start it : exten => _X.,3,AGI(My-Script.agi) The perl agi file are started without problems but i want get into this script a lot of variable: Type (SIP or IAX) src (from cdr) but that's don't work: use Asterisk::AGI; use lib "/var/lib/asterisk/agi-bin"; $AGI = new