Event response (AMI)

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When I execute the ACTION commands set then the EVENT would response back.
How would I know which ACTION are they belong/reference to? For example: ACTION: Originate
Channel: SIP/test
Exten: 215
Timeout: 30000
Context: test
Priority: 1
ActionID: 111112222333 Response: Success
ActionID: 111112222333
Message: Originate successfully queued
EVENT response when I hang up the call: Event: Hangup
Privilege: call,all
Channel: SIP/test-0000007f
Uniqueid: 1336690030.189
CallerIDNum:
CallerIDName:

Cause: 16
Cause-txt: Normal Clearing
As you can see, how would I know…

Asterisk Users 3.3 years ago 0 Answers

Event response (AMI)

Report
Question

When I execute the Action commands set then the Event would response back.
How would I know which Action are they belong/reference to? For example: ACTION: Originate
Channel: SIP/test
Exten: 215
Timeout: 30000
Context: test
Priority: 1
ActionID: 111112222333 Response: Success
ActionID: 111112222333
Message: Originate successfully queued
Event response when I hang up the call: Event: Hangup
Privilege: call,all
Channel: SIP/test-0000007f
Uniqueid: 1336690030.189
CallerIDNum:
CallerIDName:

Cause: 16
Cause-txt: Normal Clearing
As you can see, how would I know…

Asterisk Users 3.3 years ago 4 Answers

Why do I get call twice in one go?

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I understand why do I get call twice to my mobile when I execute the
following AMI command sets: ACTION: Originate
Channel: Local/800@test
Timeout: 60000
Priority: 1 and my dialplan look like this: [test]
exten => 800,1,DIAL(SIP/447xxxxxx@voip);
exten => 800,n,Hangup()
How to prevent getting called twice in one go when I execute this AMI
command? Thanks...

Asterisk Users 3.3 years ago 0 Answers

Asterisk 1.8 Transfer CallerID

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Hello, when a call comes in and is answered by colleague A, this colleague A
sees the CallerID of the external calling number. When colleague A transfers the call to colleague B, attended or
unattended, then colleague B sees the number of colleague A on his
screen while talking to the external calling number. I expect here that colleague B would see the external calling number on
the screen of his IP-phone. How can I get this behaviour ?
Thanks.
Jonas.

Asterisk Users 3.3 years ago 6 Answers

Why SendDTMF is not working?

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Am 06.05.2012 13:46, schrieb Shahid H:
> Hello,
>
> I am having a problem with SendDTMF - it is not sending the numbers
> properly during the phone call.. I want the numbers key to to be
> pressed/sent automatically after 3 seconds during a phone call. Log the actual DTMF to your console, set in logger.conf: console => something,something,dtmf
^^^^ Then try again and check if you see the actual DTMF. If you do and it
still doesn't work, try dtmfmode=inband for your voipms peer. rfc2833 has been working always unreliable…

Asterisk Users 3.3 years ago 6 Answers

Problem with SendDTMF

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Hello, I am having a problem with SendDTMF - it is not sending the numbers
properly during the phone call.. I want the numbers key to to be
pressed/sent automatically after 3 seconds during a phone call. I use software phone to test it... when I dialed 501, I cant hear anything
for about 10 seconds (this is because of SendDTMF) and then I can hear
the operator saying to enter the numbers but SendDTMF already did it?! Asterisk server are connected to voip.ms provider. I have spent many hours trying to get to work, how…

Asterisk Users 3.3 years ago 1 Answer

Asterisk Capacity

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Hello, We are currently working on a project where using .call file on asterisk
spool, outbound calls will be made from a pri line and a voice clip will be
played. We know that pri has a capacity of handling only 30 channels at a time.
Therefore, my worry is what happens if we write 100 files at a time on the
spool. Will asterisk manage the queue or how exactly will it behave. Regards, Ashish

Asterisk Users 3.3 years ago 14 Answers

Asterisk AMI SIP channel detect phone ringing

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Hey guys, I am using a SIP trunk to make outgoing calls. Outgoing calls are
going through okay. I am using the AMI to Originate a call. The
channel is not returning any event when the phone on the PSTN is
ringing. How can i detect the phone ringing on the SIP channel? Am desperate. Thanks.

Asterisk Users 3.3 years ago 0 Answers

hangup problem on T1 span

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Hello all, I'm trying to solve a problem on a T1 span setup wherein calls are
apparently not hanging up properly. The system in question is using a Xorcom Astribank with 1 full and 1
partial T1 span, and running Asterisk 1.4.36. The symptom is that when a call hangs up on a DAHDI channel (according to
Asterisk), and another outgoing call tries to open a new channel on the
same line as the hung-up call within approximately a minute of the hangup,
the new call gets a congestion notice ("all circuits busy") from

Asterisk Users 3.3 years ago 1 Answer

asterisk go to "holiday" extension but hoiday is not defined

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When a call comes in asterisk is forwarding the call to holiday extension, even though the holiday is not defined. Here is my dial plan exten => 4,1,GotoIfTime(*,*,1,jan?holiday,s,1) ; new years day
exten => 4,n,GotoIfTime(*,*,6,apr?holiday,s,1) ; easter holiday
exten => 4,n,GotoIfTime(*,*,23,may?holiday,s,1)
exten => 4,n,GotoIfTime(*,*,1,|jul?holiday,s,1) ; canada day
exten => 4,n,GotoIfTime(*,*,1,aug?holiday,s,1) ; long weekend
... Today is May 1, so why is it going to "holiday" extension?
Is there another dial plan that holidays are defined?
I'm using asterisk 1.8.10

Asterisk Users 3.3 years ago 7 Answers