similar to: Playing background message

Hellooo... Is it possible to fill a variable of PHP-based-AGI-script from dialed extension ? This is what I need to achieve: If someone dial an extension, say 777, I want the dialed extension (777) be filled into PHP variable. I need the dialed extension become a condition of PHP script. Help please... Thanks romsun _________________________________________________________ This mail sent

Hello all, As I'm sure is pretty common, I have some extensions that dial mobile numbers after a local timeout. I would like to prompt the caller to record their name after the local timeout and have the recipient be able to hear the name prior to accepting the call. Recording the message is easy enough, so I thought about doing something like dumping them into MeetMe after they record

I wish 'A(x)' was available with AgentCallBackLogin!! :( -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Matteo Brancaleoni Sent: Tuesday, February 03, 2004 6:48 AM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Playing announcement to called user prior toConfirmation show application

I am after using a web crm system which has a button to then get asterisk to dial the contact. For this I was looking at call files, which appear good for the job, I have one small problem with them though. 1/ file is created 2/ external number is called 3/ the external party answers 4/ the external party now hears ringing as you extension is now being called - bad! What I would like to

Hi, All Is there a provision for "AbsoluteTimeout" application to notify called and calling party of the reason why the call suddenly ended? This way, the parties will be much better informed, hence they will/should not think that their VOIP/telco provider(s) are providing bad service. Ta SJ

ser forwards a sip message with extension 99999996 to asterisk which plays my 'userisoffline' message and hangs up and should stop here but instead asterisk continues to process the match everything extension ._ and dials out which is not what I want... if I change the starting priority of the Dial app to a higher level than 3 asterisk stops after the hangup but then doesn't accept

Has anyone here implemented "Ring back when free" in Asterisk? The way it works in the UK is as follows: 1. A calls B. B is engaged (busy). 2. A hears "The number you called is busy. To use ringback, press 5" 3. A presses 5, and hears "Your ringback request has been accepted". 4. A hangs up. 5. Later, B hangs up. The system then calls A (if A is now busy, it

well .. good news :) i've just put in txgain=1.0 rxgain=1.0 in my zapata.conf and upgraded the Grandstream Budgettones i'm using to version 81 of the software and all seems fine .. there is still an echo but after the first couple of seconds of call it vanishes, as the echocancelling kicks in .. so far my client is happy :) now .. i have one slight problem left .. although most of my

Hi, It would be nice if it were easier for relative novices to build whole-program bitcodes for large, complex applications with hairy build systems. Several readers of this list have been trying various approaches for a few months but as far as I know we haven't yet found a good general solution. Approaches that have been tried include 1) placing wrappers for the usual tools (gcc, ar, as,

I'd like to play a sound to a user I dial (via SIP) once they answer play the sound and then allow me to talk to them. The new Cisco 7960 SIP code allows to set lines to autoanswer via the speaker phone, I'd like to play a "tone" after it rings through and then talk... Any thoughts on how to do this?

Hi Terence, I believe that this is in fact similar to an approach Coverity uses (or used at one time) as a robust solution to determine what was done during a build. I can imagine that one can build a robust system following this technique, but it also seems like it might be quite a bit of work. Another possible alternative not mentioned is to teach the compiler driver (clang, most likely) to

I am just starting to deploy asterisk in our office to use as our primary phone system - we plan to use a Voicetronix OpenLine4 card as our PSTN gateway - but one thing at a time... haven't got that far yet. Currently, i'm trying simple IP to IP calls within the office using our Cisco 7960's phones running SIP. When I make a call between these two phones, the conversation is of a

The following transfer led to a crash of asterisk, without leaving a core or any utterances in messages or debug file. It looks like the zombie which was created during the MASQ-transfer was not cleaned up... But why did it start a Dial??? And... why does Asterisk die when this happens?? Thanks!!! Michiel -- Zap/32-1 answered Zap/6-1 -- Stopped music on hold on Zap/6-1 -- Starting

hi asterisk users How can i make asterisk "ignore" invalid extensions, and go on playing the background soundfile? Normally, asteriks will take the user to the invalid extension if the caller presses anything other than 1 or 2 in the following context:: [example] exten => s,1,Answer() exten => s,2,Background(hello-world) exten => s,n,Goto(s,2) exten =>

I am interested to learn if I need to have ztdummy installed if I do not have any zaptel hardware in my machine? I have found a lot of references with RTP problems which were related to RTP timing (or lack of it). My problem is that voice coming from SIP hardware is OK, but voice going from asterisk to SIP hardware is choppy, full of noise or completely cut-off. Am I going to solve my problem

Hi, okay heres what I want to do .. simple ivr, we take a call, answer it, play a menu, dial out based on options. No problems so far. The CDR always shows the call as answered as I answer the 1st leg to play the prompts, I am actually more interested in if the 2nd leg - the outbound part - has been answered or not before the call is hungup. How can I get this and record the information in

<P>hi,</P> <P>Can anybody pls tell me how to activate loggin CDR on mysql db. I tried&nbsp;editing the /etc/asterisk/cdr_mysql.conf file and recompiling asterisk, it didn't work. Normal loggin on Master.csv file works fine.</P> <P>Thank you inadvance,</P> <P>Surajee</P> <P>&nbsp;</P><br> --------------This mail sent

Hi. Has anyone experienced with the new firmware .77 ? There's Day Light Saving time now, but haven't time to play with it, till now. Matteo. -- Matteo Brancaleoni Espia System Administrator - IT services Website : http://www.espia.it Email : mbrancaleoni@espia.it

Is it possible to play and audio file into a meetme conference for both parties to hear? I thought I remembered reading something about it, but I can't find it now. Any help would be greatly appreciated. Paul

Hi All, I am in desperate need of this feature. I want to play background music during a call while the 2 parties are having some lovely conversation (or maybe give them a sort of cursing background if they are cursing each other). I found this post which talks about creating a ghost call with the help of queues and putting that queue in a meetme room where queue will play the song/curse and the