I am using 1.4.43 currently. I am using the AMI to originate a call over a SIP Trunk to my cell XXX506YYYY. works fine. when the call is active I do a "core show channels concise" and I get: SIP/testsystem-00000ad0!smvoice-dialout!callprogress!4!Up!AGI!smvoice!0!!3!24!(None) My AGI is called smvoice. No place does my number show up. How do I "lookup" my call so I can "hangup" the call at a later time. In my case there my be more than one call active at a time, and I want to hangup the correct call. I know I need the data "testsystem-00000ad0" to cancel my call but how do I "associate" that with my number so I can find the right call to hangup. Thanks, Jerry
Since you're using sip, use sip show channels and pick the call-id from there. -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Jerry Geis Sent: Wednesday, November 07, 2012 7:27 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] how to lookup a call I am using 1.4.43 currently. I am using the AMI to originate a call over a SIP Trunk to my cell XXX506YYYY. works fine. when the call is active I do a "core show channels concise" and I get: SIP/testsystem-00000ad0!smvoice-dialout!callprogress!4!Up!AGI!smvoice!0!!3!2 4!(None) My AGI is called smvoice. No place does my number show up. How do I "lookup" my call so I can "hangup" the call at a later time. In my case there my be more than one call active at a time, and I want to hangup the correct call. I know I need the data "testsystem-00000ad0" to cancel my call but how do I "associate" that with my number so I can find the right call to hangup. 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
On Wed, Nov 7, 2012 at 7:27 AM, Jerry Geis <geisj at pagestation.com> wrote:> I am using 1.4.43 currently. > > I am using the AMI to originate a call over a SIP Trunk to my cell > XXX506YYYY. works fine. > when the call is active I do a "core show channels concise" and I get: ><snip>> How do I "lookup" my call so I can "hangup" the call at a later time. > >Since you're using AMI to originate the calls, you should then also be able to add an ActionID to the originate command. You should then be able to lookup the call in AMI using the ActionID. -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com <http://www.selbytech.com> -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20121107/815111cc/attachment.htm>
I would not know if this is something that can be helpful to you, but in WombatDialer we associate a channel variable to with an unique-id to each call, so that we can reattach to a set of calls if the AMI connection goes down and we can be absolutely sure that what we are looking at is the call we think it is. It is not really expensive to do - just a GetVar per channel to mek sure our assumptions are correct. 2012/11/7 Jerry Geis <geisj at pagestation.com>> I am using 1.4.43 currently. > > I am using the AMI to originate a call over a SIP Trunk to my cell > XXX506YYYY. works fine. > when the call is active I do a "core show channels concise" and I get: > > SIP/testsystem-00000ad0!**smvoice-dialout!callprogress!** > 4!Up!AGI!smvoice!0!!3!24!(**None) > > My AGI is called smvoice. > No place does my number show up. > How do I "lookup" my call so I can "hangup" the call at a later time. > > In my case there my be more than one call active at a time, and I want to > hangup the correct call. I know I need the data "testsystem-00000ad0" to > cancel my call > but how do I "associate" that with my number so I can find the right call > to hangup. > > 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> >-- Loway - home of QueueMetrics - http://queuemetrics.com Test-drive WombatDialer beta @ http://wombatdialer.com -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20121112/a3d10664/attachment-0001.htm>