asterisk users - Jan 2013 - question on SIP trunk and AMI to place call

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-0000000d
ChannelState: 0
ChannelStateDesc: Down
CallerIDNum:
CallerIDName:
AccountCode:
Exten:
Context: testmachine
Uniqueid: 1359035395.20

In this event or any event following I do not see
the phone number that I dialled. How do I "correlate"
the "SIP/testmachine-0000000d" to the number I just dialed????
(purpose is to hangup the call later if I need to interrupt it)

Now if I am using a machine with actual hardware cards, the phone
number is included as part of the Channel so I can look that up.
but for a SIP trunk the phone number dialled does not come over the AMI.

How do I match up the call I just started (using AMI over SIP trunk) to 
the number I called?

Thanks,

jerry

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 <geisj at pagestation.com> 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-0000000d
> ChannelState: 0
> ChannelStateDesc: Down
> CallerIDNum:
> CallerIDName:
> AccountCode:
> Exten:
> Context: testmachine
> Uniqueid: 1359035395.20
>
> In this event or any event following I do not see
> the phone number that I dialled. How do I "correlate"
> the "SIP/testmachine-0000000d" to the number I just dialed????
> (purpose is to hangup the call later if I need to interrupt it)
>
> Now if I am using a machine with actual hardware cards, the phone
> number is included as part of the Channel so I can look that up.
> but for a SIP trunk the phone number dialled does not come over the AMI.
>
> How do I match up the call I just started (using AMI over SIP trunk) to
> the number I called?
>
> Thanks,
>
> jerry
>
>
>
> --
> ______________________________**______________________________**_________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>               http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>  
http://lists.digium.com/**mailman/listinfo/asterisk-**users<http://lists.digium.com/mailman/listinfo/asterisk-users>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
<http://lists.digium.com/pipermail/asterisk-users/attachments/20130124/c05aab6b/attachment.htm>

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-0000000d 

And get the dialed number from that.

Actually you could issue the AMI command core show channels verbose.

 

From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces
at 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 <geisj at pagestation.com> 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-0000000d
ChannelState: 0
ChannelStateDesc: Down
CallerIDNum:
CallerIDName:
AccountCode:
Exten:
Context: testmachine
Uniqueid: 1359035395.20

In this event or any event following I do not see
the phone number that I dialled. How do I "correlate"
the "SIP/testmachine-0000000d" to the number I just dialed????
(purpose is to hangup the call later if I need to interrupt it)

Now if I am using a machine with actual hardware cards, the phone
number is included as part of the Channel so I can look that up.
but for a SIP trunk the phone number dialled does not come over the AMI.

How do I match up the call I just started (using AMI over SIP trunk) to the
number I called?

Thanks,

jerry



--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
              http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL:
<http://lists.digium.com/pipermail/asterisk-users/attachments/20130124/4b95aa9d/attachment-0001.htm>

> ------------------------------------------------------------------------
> 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
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
<http://lists.digium.com/pipermail/asterisk-users/attachments/20130124/0e2e3c29/attachment.htm>

> 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-0000000d
>
> And get the dialed number from that.
>
> Actually you could issue the AMI command core show channels verbose.
there is no core show channels verbose on Asterisk 11. There is on 
asterisk 1.4,

core show channels on asterisk 11 has been changed.

jerry
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
<http://lists.digium.com/pipermail/asterisk-users/attachments/20130124/c259ccab/attachment.htm>

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-0000000d
> ChannelState: 0
> ChannelStateDesc: Down
> CallerIDNum:
> CallerIDName:
> AccountCode:
> Exten:
> Context: testmachine
> Uniqueid: 1359035395.20
> 
> In this event or any event following I do not see
> the phone number that I dialled. How do I "correlate"
> the "SIP/testmachine-0000000d" to the number I just dialed????
> (purpose is to hangup the call later if I need to interrupt it)
> 
> Now if I am using a machine with actual hardware cards, the phone
> number is included as part of the Channel so I can look that up.
> but for a SIP trunk the phone number dialled does not come over the AMI.
> 
> How do I match up the call I just started (using AMI over SIP trunk) to
> the number I called?
> 
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 contain the channel name being dialled, and
the Dialstring: field will contain the non-technology specific portion
of the thing being dialled.

For example:

exten => 500,1,Dial(SIP/digium02)

Results in:

Event: Dial
Privilege: call,all
SubEvent: Begin
Channel: SIP/10.x.x.x-00000002
Destination: SIP/digium02-00000003
CallerIDNum: 657-5309
CallerIDName: digium01
ConnectedLineNum: <unknown>
ConnectedLineName: <unknown>
UniqueID: Asterisk-01-1359052866.2
DestUniqueID: Asterisk-01-1359052866.3
Dialstring: digium02

-- 
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org

>
>
> 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 contain the channel name being dialled, and
> the Dialstring: field will contain the non-technology specific portion
> of the thing being dialled.
I get that even on the system with the PRI card and using DAHDI
however I am not getting that event on the system with the SIP trunk .

Is there something to enable to get that???
Both systems are running Asterisk 11.0.2.

Thanks,

Jerry
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
<http://lists.digium.com/pipermail/asterisk-users/attachments/20130124/a3f2f7c8/attachment.htm>

This might have changed but IIRC /etc/asterisk/manager.conf controls what
events you have access to.

 

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Jerry Geis
Sent: Thursday, January 24, 2013 1:14 PM
To: Asterisk Users Mailing List - 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+ManagerEvent_Dial
 
Note that the Channel: field will contain the name initiating the Dial,
the Destination: field will contain the channel name being dialled, and
the Dialstring: field will contain the non-technology specific portion
of the thing being dialled.

I get that even on the system with the PRI card and using DAHDI
however I am not getting that event on the system with the SIP trunk .

Is there something to enable to get that???
Both systems are running Asterisk 11.0.2.

Thanks,

Jerry

-------------- next part --------------
An HTML attachment was scrubbed...
URL:
<http://lists.digium.com/pipermail/asterisk-users/attachments/20130124/7f69d5c2/attachment.htm>