similar to: Java Asterisk Event Message

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

The Asterisk Development Team has announced the release of Asterisk 11.9.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 11.9.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: Bugs

The Asterisk Development Team has announced the release of Asterisk 11.9.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 11.9.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: Bugs

Hi all, Now i want to check a number (channel) online, offline or unreachable on asterisk but i don`t know to do. Can anyone help me to solve this issue. Thanks and best regard! -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110109/c193b48d/attachment.html>

Hi all, I want to be able to see the status of my Agents on a web interface. I have no idea how to do so. I have found a few sample script to communicate with queues manager to view queues.But I couldn't find any on viewing the agent status. Could anybody give me a clue? Regards, Pim

I am trying to get updateconfig working. I found an example of updating configuration files here: http://www.voip-info.org/wiki/index.php?page=Asterisk+Manager+API+Action+Upd ateConfig When I tried it the conf file was updated but the new entry was not added. action:updateconfig reload:no srcfilename:manager.conf dstfilename:manager.conf Action-000000:append Cat-000000:newuser

I'm curious (ok, well I admit it - it's for perosnal gain) what methods people are using to manage asterisk config files when they have multiple asterisk systems? Some sort of revision control such as cvs,rcs or subversion? A central 'config server' where you edit the files and then rsync them out? I have 5 systems to manage, and it seems that about the only common file is

Hi, I have questions about the Polycom 601 and side card.... 1) In the side card the lights all time off... But all functions it's ok. I need help with extension module of polycom... All works fine... But lights not work.... So... I don't know when any person or extension is busy... Any ideas? , Olger On 3/27/06 11:34 PM, "asterisk-users-request@lists.digium.com"

Hi All My dial plan has the following context: [sip-guest] exten => _!.,1, Answer exten => _!.,n, verbose(1,[${EXTEN}@${CONTEXT}]) exten => _!.,n, resetcdr(w) exten => _!.,n, resetcdr(w) exten => _!.,n, set(DNIS=${EXTEN}) exten => _!.,n, resetcdr(w) exten => _!.,n,

I would like to play audio--using controlplayback-- to 2 channels--agent and caller- simultaneously. Tried meetme,confbridge,originate without success. Tried redirecting the channels to a context, playing audio to the agent's channel and then bridging the 2 channels. The problem with this is as soon as the bridge is created the audio stops. I can provide the dialplan details, if anyone is

There is one more thing to try: http://snapvoip.blogspot.com/2009/07/appkonference-asterikast-high.html I would appreciate if anyone can comment on the feasibility of playing an audio file to the caller and callee using ControlPlayBack and appkonference. Much of the reviews indicate that appkonference is an over-kill for an audio as its main functionality is with video. Going past that. Thanks

Hi All, Downloaded latest version of Asterisk from www.asteriskwin32.com and installed on Windows 7. Here is my sip.conf [general]context = demo ; Default context for incoming callsbindport = 5060 ; UDP Port to bind to (SIP standard port is 5060)bindaddr = 0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)srvlookup = yes ; Enable DNS SRV

To: asterisk-users at lists.digium.com From: webaccounts173 at jgoettgens.de Date: Wed, 29 Jul 2015 16:11:31 +0200 Subject: Re: [asterisk-users] Windows Asterisk Help Downloaded latest version of Asterisk from www.asteriskwin32.com and installed on Windows 7. Here is my sip.conf

Hi All Has anyone used Asterisk for a Call Center operation? What I mean is: given a list of phone numbers, can Asterisk dial each number, play a message and accept some DTMF? I ask because I am an employee of a non-profit company based in San Diego, CA. I already evaluated Voicent and Voxeo. The former has expensive licensing terms and the latter is not best suited for a call center. I would

Hi All I am trying to dial out using SIP and Vonage using the instructions : <a href="http&#58;&#47;&#47;www.voip-info.org&#47;wiki&#47;view&#47;Asterisk&#43;and&#43;Vonage" target="_blank"