similar to: How do you handle queues with AMI?

Dear all, Anyone of you knows how to play an audio file at the beginning of a call for both Caller and Callee? A(x) of Dial application only plays audio for callee. I don't want to use MeetMe because I want to use Monitor and MixMonitor. Thank you! ________________________________ Este mensaje se dirige exclusivamente a su destinatario. Puede consultar nuestra pol?tica de env?o y recepci?n

Hi All, I want to track a call that is originated using originate AMI command through AstManProxy server. I m using AstManProxy server and I developed an AstManProxy client. By using my AstManClient program I can able to login AstManProxy server. Now I can able to issue/send originate command to generate a call but I m very confuse that I cannot able to track my call. The AMI events were

Is there any way to do this? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120307/77764e4b/attachment.htm>

Hello User-List, is it possible to play a sound file directly to a caller channel? Like this AMI command Action: Originate Channel: SIP/20-00001d41 Application: Playback Data: /path/to/audio/file I get an Error Message. My intension is to play a sound file to a caller and the other callers don't hear this. Can someone help me ? Thanks a lot Bye Daniel

Hi all, How can I playback a file within an active call? I've tried with ChanSpy whisper mode like this (using AMI): Action: Originate Channel: Local/9999 at default Priority: 0 Variable: MSG=test Application: ChanSpy Data: SIP/1234-123 Async: 1 and in the dialplan: [default] exten => 9999,1,Answer() exten => 9999,n,Wait(2) exten => 9999,n,Playback(${MSG}) Where

Is there a way to inject sound from a sound file into an established call using AMI? I have an established call from which I can record either or both legs. I can additionally "spy" on the call. Is there any way I can play a sound file into the call and not loose the ability for the people to continue talking while listening to the sound file? -- Jim Dickenson mailto:dickenson at

Hi all, I am trying to make decisions in my dialplan based on {DIALSTATUS}. I am creating calls using AMI (rawman with parameters via URL) with action:Originate. I am using SIP and an outside voip provider for the calls. If I define the number to call in the Channel parameter (e.g. SIP/15555555555 at myvoipprovider, the call gets placed before entering the context that I defined. I understand

Hi, I'm writing a Java client/server application that talks to the Asterisk manager interface via the asterisk-java stuff. The idea being it will give you an app to run on your desktop that monitors your phone essentially. Once I've got something vaguely working it will be released under the GPL and hopefully people will contribute to it etc... As part of this, I'm currently

hi list, i have a question about the args of queue: when we use Queue() app, there are some arguments than can use. help from CLI: Queue(queuename[,options[,URL[,announceoverride[,timeout[,AGI[,macro[,gosub[,rule]]]]]]]]) well.. i'm trying to identify who has taken the call on a queue, and, when agent conected, launch a macro with some args based on who takes the call i do: exten =>

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

Hello, I have asterisk 1.6.0.20 and Is it possible to add Reason header on Hangup: Reason: q.850;cause=17 Thanks -- Best Regards, Giedrius -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100221/d29c02b8/attachment.htm

Is there any AMI action that logs a realtime agent? I mean, if you send it, queue_log and queue_member get the corresponding inserts. Regards Sebastian -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090312/68bb3a3a/attachment.htm

Folks, I am strugging with Asterisk 1.4 Vs 1.6 differences over AMI Originate? Here is the pastebin... http://pastebin.ca/1805594 Not sure why the local channel won't send to context while the remote channel does. Worked fine in 1.4 but 1.6.1 has issues. Any help? Ritesh -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I'm executing some commands using AMI... I suppose the log is saved in some place, but I don't know where... where is it saved?More details: I'm executing a UpdateConfig in the voicemail.conf file, but the file is not updated, so I would like to know why...Thanks, Anahi Anahi Ludue?a _________________________________________________________________ Ahora

Hi guys I seem to be having a problem, I don't know if it's a bug or whether I'm just doing it incorrectly. I want to set about 3 channel variables when I originate a call via AMI. All the documentation I have found says to do it like this: Variable: variable1=value|variable2=value|variable3=value However when I do this it runs them all together and I end up with:

Hello! I want to thank everyone who helped me out with tips for load balancing asterisk machines in a cluster. I have encountered a new problem that is related to SIP diversion headers in the INVITE. I make calls through the manager interface and now want to add a SIP-Diversion header that changes the CallerID of a number that is not available on the trunk, the CallerID to be visible externally

In one of my center , its not taking root password. Anyways to recover it ? In other terms , I lost the control of server. Any solution or re-installation is the only way left ? I am using CentOS. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090122/ef95ad6e/attachment.htm

Hi All, pl help me with this basic question. I have a users (soft clients) with usernames having Alphabetics. I want to use Asterisk as my server. How should I have the dial plans as there are no numbers involved . so How can I make the configuration to work ( with numbers I can get this done using extensions.conf) my expected result is : alice at pbx.com should be able to call bob at

Jim, and all: Thanks for the response. If I can repeat what you are saying: you don't have to define the multiple lines in sip.conf? For comparison, with my current esi setup, we have 10 outgoing lines. If one line is busy, then the service just rolls to the next number. This is set up with the phone service. That doesn't have to done with outgoing sip lines? Does the dialstatus

if i run let's say 1 pbx running on my main linux box and a another on my windows box if a person dial my main number and press lets say 1 are it possible to transfer the call over to my other pbx hope anyone understand -------------- next part -------------- An HTML attachment was scrubbed... URL: