Displaying 20 results from an estimated 5000 matches similar to: "How do you handle queues with AMI?"
2011 Dec 15
3
Play audio file for both Caller and Callee in a call
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!
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2010 Aug 08
3
How to track a call result originated from originate AMI command
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
2012 Mar 07
1
Finish ChanSpy() when channel spied hangs up
Is there any way to do this?
Thanks
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2010 May 16
1
play a sound file directly to a caller channel
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
2010 Aug 10
1
Playback during call
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
2008 Nov 20
1
Playback using AMI
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
2009 Jun 03
1
Using DIALSTATUS question
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
2006 Apr 13
1
AgentCalled event
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
2009 May 07
1
Macro arguments on app_queue
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 =>
2008 Oct 14
1
Help With AMI
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
2010 Feb 21
2
add Reason header on hangup
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
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2009 Mar 12
1
Queue Realtime agents LOGIN for ami
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
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2010 Feb 22
1
AMI Originate differences between 1.4 and 1.6.1
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
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2010 Mar 05
1
AMI logs
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
2009 Dec 04
2
Multiple Channel Variables with AMI Originate
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:
2011 Oct 07
2
Add SIP diversion header in originate from AMI?
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
2009 Jan 22
7
Root Password not taking
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.
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2010 Apr 28
6
Dial plan question.
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
2010 May 10
1
More clarification on outbound sip channels.
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
2011 Nov 03
1
2 pbxes
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
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