similar to: API or Call command

Hi We run a hosted VoIP service for multiple customers off the same server and I'm having an odd issue with just one customer in particular. We're using realtime in a MySQL DB and this is their dialplan *************************** 1. row *************************** context: pcsu-Identifier exten: s priority: 1 app: Answer appdata: *************************** 2. row

Hi, I am working on a project to record agent calls when completing specific transactions with customers. Since these calls do not go through the asterisk box (They go through a lucent G3), we're thinking that service observe would be the easiest way to accomplish our goal. Here's what I need: On demand, I need to be able to attach to the switch, dial the service observe code, make

Greetings to All. A little background about what I am trying to do, and please excuse the length of this post. I am setting up a voicemail (VM) system based on Asterisk. From what I've heard Vonage uses Asterisk as their VM platform as well. I am running 1.2beta with a MYSQL backend for extensions and VM user info. All the sound files and vm messages are being stored through an NFS

I'm running Asterisk CVS-v1-0-12/20/04. I'm using PHP with Manager API Here is the code: #################################################################### # Make call #################################################################### $socket = fsockopen($ask_db,"5038", $errno, $errstr, $timeout); if (!$socket) { echo "$errstr ($errno)<br /\n"; } else {

I have got time-limited calls with a warning announcement working now ..... almost. Something is going slightly wrong, because the call file that gets created to trigger the announcement is persisting even after both ends of the call have hung up. When a code is dialled, the user enters the context "time-limited-call". An announcement is played and the user then dials the number

sorry for the misunderstanding, robert! the point is: during the caller is listening to the soundfile played to him the dialplan should continue to dial the sip phone 100 and after sip phone 100 is answered and the announcement file is played the caller should be connected to the sip phone 100. the behaviour is just like MoH, but the problem is, that the caller has to hear a soundfile from the

With the 1U Apple G5 server at a good reliability/storage/pricepoint level... got to thinking about compiling Asterisk on OS X.. at least for SIP phone call switching, voicemail, etc. Has anybody attempted this? Email me off list if this is too dev-heavy for the user list. Thanks, Ted W -----Original Message----- From: asterisk-users-request@lists.digium.com

You can send both paties to a meetme conference with Manager Redirect. Or if you are feeling more adventurous you could load the Manager Bridge patch that I posted to the bugtracker two months ago. It allows bridging of any two existing channels together through a manager action: http://bugs.digium.com/view.php?id=4297 MATT--- -----Original Message----- From: Roland Zagler

Hello all, I?ve found another issue with the queue application. Assuming I?ve configured a queue with a long periodic announcement and have two queue members assigned. Both queue members are busy at a time, while another caller is joining the queue. After a while the periodic announcement is played back to the caller, in that case it takes about 40 seconds to be played back. If then one of the

Folks, Apologies ahead of time if this has already been asked (read the list for the last month looking for something similar). I have been trying to get the Background command to work with no joy yet. Here is what I am trying to do: 1. Answer the call. 2. Play the message in the background, while waiting on DTMF from user. 3. If I get a "1", then interrupt the message and dial the

When using the callback feature on agents I notice that when the queue calls one of the agents and the agent picks up the call they hear nothing until pressing the # to accept the call. Only then does my announcement play back to the agent after which the call is immediately connected. Is there a way to have the announcement played to the agent before they press # to accept. I have ackcall=yes

Hi All I'm tracing a very strange problem which I could reproduce with different versions up to 1.2.5 (sorry, didn't update to a newer one yet). Scenario 1: Problem does not occure. ============================= Sip Phone registered directly to the Asterisk. exten => i,1,Zapateller() exten => i,n,Playback(invalid,noanswer) exten => i,n,Hangup() Works like expected. I dial an

Hi, I have Asterisk-1.2.18 install with FreePBX & more than 75 extnsion, daily I come accross an issue & try resolving them its either user learning curve or my ignorance. But, I dont know what to say regarding this issue. I have my Dial Plan for internal users to have a 3 Digit Extensions. So instance my Ext is 239 & someone dials the main #, its gets the

I'm testing a few IAX trunk scenarios in a controlled lab. From server2 extension 5000 (Server IP Address 172.16.200.212) I dial 6001 which goes across the IAX trunk to server 1 (IP address 172.16.200.210). Instead of ringing the 6001 phone, it plays tt-weasels (the s extension). When I dial 6099 it also plays tt-weasels as it's supposed to, but it's not the tt-weasels under its

Maybe try progress() instead of answer () ?????? 23 ????? 2016 7:19 PM,? "Jean Aunis" <jean.aunis at prescom.fr> ???: > Thank you, I just tried your suggestion. Strangely, the announcement is > played only if I try to dial a SIP peer which is not available (not > registered to be more precise). If the SIP peer is available, I only get > the ring tone, and never hear

Has anyone created like a broadcast dialplan, if so care to share it. What I'd like to do is create an extension so when someone calls that extension they can leave a voicemail. Right after it is recorded, I need that voicemail to played on all phones on that system... E.g.: 1) Administrator --> Dial special number 2) Record emergency message (e.g. Snow day don't come in) 3) Hang up 4)

Hi, I'm trying to do the following but can't quite get it right: 1) Callers rings DID number 2) Asterisk rings the appropriate channel for 30 second, if no answer sends to voicemail (no problem up to here, of course) 3) IF the channel is answered Asterisk plays an audio file 4) Asterisk connects caller with me I need to do this to "cover up" the delay within the first few

Hi, I'm trying to do the following but can't quite get it right: 1) Callers rings DID number 2) Asterisk rings the appropriate channel for 30 second, if no answer sends to voicemail (no problem up to here, of course) 3) IF the channel is answered Asterisk plays an audio file 4) Asterisk connects caller with me I need to do this to "cover up" the delay within the first few

Hello, We are working on a new Asterisk installation and have run into some problems related to playing MusicOnHold for a caller when they have been placed on hold by an agent, that took the call from a queue. A. When pressing the HOLD button on SNOM 190 and Grandstream BudgeTone SIP phones, MusicOnHold works fine when making inbound or outbound direct calls by extension. Music starts to play

Hi, I have 6 Red FXO with TDM2400p in my PC. I have install asterisk and dahdi driver. Scenario is jitsi-----> asterisk server-----> analog PBX ----> landline phone I configured this scenario as follow in chan_dahdi.conf file ; General options [channels] usecallerid=yes hidecallerid=no callwaiting=yes threewaycalling=yes transfer=yes