I just put a break at dial_exec_full (app/app_dial.c for Asterisk 11.0.2)
did my AMI call
Action: Originate
Async: yes
Channel: SIP/testsystem/XXX
(calls from my machine over SIP trunk to another 11.0.2 box that has
a PRI card to make a call out to my cell)
and did not get a break.
Why is
The Dial events are created by app_dial. So long as you are using
app_dial to create your outbound channel, you should have that event.
Channel technology shouldn't matter.
I am using the same AMI method to start both calls.
Action: Originate
Channel: DAHDI/18/XX
or
Action: Originate
Chann
On 01/24/2013 01:13 PM, Jerry Geis wrote:
>>
>>
>> You probably want the Dial event. It is raised both at the beginning of
>> the Dial, as well as when the Dial completes.
>>
>> https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+ManagerEvent_Dial
>>
>> Note that the Channel: field will contain
- Non-Commercial Discussion
Subject: Re: [asterisk-users] question on SIP trunk and AMI to place call
You probably want the Dial event. It is raised both at the beginning of
the Dial, as well as when the Dial completes.
https://wiki.asterisk.org/wiki/display/AST/Asterisk+11
You probably want the Dial event. It is raised both at the beginning of
the Dial, as well as when the Dial completes.
https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+ManagerEvent_Dial
Note that the Channel: field will contain the name initiating the Dial,
the Destination: field will con
On 01/24/2013 10:46 AM, Jerry Geis wrote:
> When I am monitoring the AMI I see the following event
> for a call I just made over a SIP trunk.
>
> Event: Newchannel
> Privilege: call,all
> Channel: SIP/testmachine-000d
> ChannelState: 0
> ChannelStateDesc: Down
> CallerIDNum:
> CallerIDName:
>
Not the greatest solution, but since you are most likely using a script for the
AMI process, you could do an
Asterisk --rx "core show channels verbose"|grep SIP/testmachine-000d
And get the dialed number from that.
Actually you could issue the AMI command core show channels verbose.
there
Have you tried and looked up all events generated when you place the call?
some of them are bound to have the variable callerid set
yes I have looked at all of them, CallerID is not set to the number I am
calling.
Jerry
-
.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tiago Geada
Sent: Thursday, January 24, 2013 11:24 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] question on SIP trunk and AMI to place call
Have you tried and looked up all events generated when you place the call?
some of them are bound to have the variable callerid set
On 24 January 2013 16:46, Jerry Geis wrote:
> When I am monitoring the AMI I see the following event
> for a call I just made over a SIP trunk.
>
> Event: Newchan
When I am monitoring the AMI I see the following event
for a call I just made over a SIP trunk.
Event: Newchannel
Privilege: call,all
Channel: SIP/testmachine-000d
ChannelState: 0
ChannelStateDesc: Down
CallerIDNum:
CallerIDName:
AccountCode:
Exten:
Context: testmachine
Uniqueid: 1359035395.2
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