[asterisk-users] question on SIP trunk and AMI to place call

Tiago Geada tiago.geada at gmail.com
Thu Jan 24 11:23:38 CST 2013


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>


More information about the asterisk-users mailing list