Displaying 20 results from an estimated 800 matches similar to: "Play audio file for both Caller and Callee in a call"
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
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
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 =>
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 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:
2010 Jul 05
7
How to Dialogic 240/JCT-T1 interface with Asterisk?
Hello all Asterisk Users,
This is my first post here.
We are in a process of moving Dialogic 240/JCT-T1 from old voicemail server
to Asterisk box.
Which card drivers do we need?
Please share experience if anyone have successfully configured Dialogic
JCT-T1 card with asterisk?
Only source proves that this card work with *
http://lists.digium.com/pipermail/asterisk-dev/2003-April/000244.html
2009 Jan 28
1
Scope of variable
I have this extension:
exten => 1322,1,Answer()
exten => 1322,n,Set(CfMC_AMDValue="NotChecked")
exten => 1322,n,GotoIf($["${CfMC_DoAMD}" != "Yes"]?NOAMD)
exten => 1322,n,AMD()
exten => 1322,n,Set(CfMC_AMDValue = ${AMDSTATUS})
exten => 1322,n(NOAMD),Wait(1)
exten => 1322,n,UserEvent(E1322-1,${CfMC_ActionID}=${CHANNEL} &
${CfMC_AgentToUse}
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 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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2011 Jan 06
2
Benefit of PRI vs SIP trunk calls
Does Asterisk, currently using version 1.4, get any more information about the result of an outbound call made over a PRI line compared to a call via a SIP trunk?
As an example, in a PRI call there is this message that shows up on the console:
[2011-01-05 14:59:02] -- Channel 23 detected a CED tone from the network.
for a call to a fax machine. Does asterisk set anything that a dialplan can
2011 Apr 14
1
Microsoft Lync server and Asterisk access
We have a client that currently has a Microsoft Lync setup. I must admit I know nothing about this setup.
What we would like to be able to do is allow the phones on desks connected to this server the ability to dial something that would allow the phone to access an asterisk box to be able to do an agent login over their LAN.
Is there any way to do this? Can the Lync server have a SIP trunk to
2009 Feb 11
0
ChanSpy problem
I have an extension defined like this:
exten => do_monitor,1,Answer()
exten => do_monitor,n,NoOp(Just got '${CfMC_ActionID}')
exten => do_monitor,n,ChanSpy(${CfMC_WhoHear},qX)
exten => do_monitor,n,Hangup()
I use an AMI packet like this:
Action: Originate
Channel: Agent/1001
Exten: do_monitor
Context: cfmc_cdi_private
Priority: 1
Variable: CfMC_ActionID=callE1334
Variable:
2009 Feb 04
0
Stopping chanspy
I would like to be able to stop the chanspy application and go to the next
step in the dialplan but I do not see a way to do that.
I have looked at the code and I do not see a way to stop the chanspy
application.
Even if there are no channels that match the chanprefix pattern the chanspy
application is not exited.
Hitting the * key stops spying on a channel but then starts spying on the
same
2011 Apr 15
5
Possible bug in Hangup() (Asterisk 1.4.x)
Hello,
On an Asterisk 1.4.33.1 in a simple scenario:
[test]
exten => _X.,1,Dial(SIP/12345 at peer01,,,)
exten => i,1,Hangup(${HANGUPCAUSE})
exten => t,1,Hangup(${HANGUPCAUSE})
exten => h,1,Hangup(${HANGUPCAUSE})
I have noticed that no matter what value we set in the Hangup(<cause
code>) commands, if the call is not answered by peer01 for any reason,
the actual cause code
2009 Jun 24
3
dahdi-linux-2.2.0 compile problem
I have an i686 cpu and when compiling from source I get this error:
touch /usr/src/dahdi-linux-2.2.0/drivers/dahdi/xpp/init_fxo_modes.verified
Building modules, stage 2.
MODPOST
WARNING: could not find
/usr/src/dahdi-linux-2.2.0/drivers/dahdi/vpmadt032_loader/.vpmadt032_x86_32.
o.cmd for
/usr/src/dahdi-linux-2.2.0/drivers/dahdi/vpmadt032_loader/vpmadt032_x86_32.o
Anyone else seeing this?
2011 Jan 10
3
How to check a number online or offline
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!
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