similar to: Send message to AMI from dialplan

I understand how to control the first local channel, but an having trouble getting the second local channel to enter stasis. I setup have the following extensions.conf to handle 1000 (basically had it setup so if first stasis not there try second, but believe second channel never processes the dial plan so even if second line was hello-world2 it would not matter. [mycontext] exten =>

HI Matt, I am trying to replicate this particular problem. We are seeing more frequently where the Event: AsyncAGIExec is never being sent. The two scenarios I have seen in tests yesterday and today... We sendl an AMI action. For example, play a short file or hangup. AMI Events will indicate it did the work, but we never receive the Event: AsyncAGIExec with a result at all. Asterisk debug

Hi Team, Thanks for your great job an Asterisk new features developments. I installed asterisk-12 Beta and found some changes as well which i notice to put in-front of your knowledge, don't know that bug of new modification into objects or old version (asterisk-11) mistake corrected that time, anyway *Asterisk-12:* Array ( [Event] => ConfbridgeMute [Privilege] => call,all [Conference]

We are communicating with Asterisk via AMI. Running Asterisk version 13.18.5 on an Ubuntu box. If you look at the event response, the Result field is filled with random characters. I'm not sure what to do because that is a completely random result. It makes no sense. We send the following command to asterisk via AMI Action: AGI ActionID: C44415 Channel: SIP/192.168.40.105-00001338

Please help find the cause of strange behavior res_pjsip. Making outgoint call to other sip server (CommuniGatePro), my asterisk suddenly sends BYE after picking up! Partial log of an outgoing call with full debug is attached and on web: http://pastebin.com/tLNCpx4d No diagnostic messages why asterisk suddenly decided to hangup i don't found :( There are suggestions or strong belief

Hi I am experiencing Asterisk Crash. Log got stopped when asterisk crashed. Please help me to identify the reason and fix this issue. Asterisk: 1.8.20 I am using AMI and fastAGI to control the call. Some part of dial plan is also defined in extensions.conf I am experiencing this crash when app_meetme conference functionality is used with more than 3 parties. I faced this issue with

Hello, I am using Asterisk 11.0.1 and do not notice any changes regarding the Transfer on newer Asterisk 11.x versions. I am using AMI and controlling a channel via AsyncAGI. I send a Transfer cmd (such as the following) Action: AGI ActionID: C8 Channel: SIP/1004-00000002 CommandID: C8 Command: EXEC Transfer SIP/1003 Destination phone starts ringing. If it answers the

Hello List, Good day, We have an application, where we redirect a channel to meet me. Sometimes the channel is getting hanged up by Asterisk, and we get an hang-up event. Please reply back, if any one faced such issue. Here is the hangup event info, -HANGUP {calleridname=<unknown>, connectedlinename=<unknown>, uniqueid=1384413814.79523, cause=0,

Hello. I've experience a rather surprising behaviour of the AMI 1.1 > Event: Newstate^M > Privilege: call,all^M > Channel: SIP/XXXXXX-089c63b8^M > ChannelState: 4^M > ChannelStateDesc: Ring^M > CallerIDNum: XXXXXXXX^M > CallerIDName: YYYYYYYYY^M > Uniqueid: 1256089773.59^M Usually ChannelStateDesc gives me 'Ringing' but sometimes it only gives me

There''s got to be an easier way... I am building an event manager and as part of it, the user selects the time zone of the event from a drop-down list using the time_zone_select helper. This is stored in the events table as a string. Because of legacy data which is stored in UTC, I want to keep the dates and times in the table as UTC but when I display it, I want to show the abbreviated

Hello, I'm using 1.6.1-rc4. When sending a userevent, (with "UserEvent(MyEvent);" in extensions.ael), I've got : Event: UserEvent Privilege: user,all UserEvent: MyEvent I can't see any Uniqueid field as mentioned http://www.voip-info.org/wiki/view/Asterisk+cmd+UserEvent or http://www.the-asterisk-book.com/unstable/applikationen-userevent.html Is this Uniqueid mandatory ?

On Wed, Aug 15, 2007 at 05:29:03AM -0000, puppet wrote: > #761: mount tries to mount an already mounted FS > -----------------------------+---------------------------------------------- > Reporter: porridge | Owner: luke > Type: defect | Status: new > Priority: normal | Milestone: misspiggy >

I use Jack for getting callee sound. Dial with option M(): JACK_HOOK(manipulate,i(rec_737219:input),o(rec_737219:output),c(rec_737219))=on This works fine, but I need to connect the sound channel to Jack *before* the actual answer. As you can see in the AMI log, between "Ringing" to JACK_HOOK there is a 6 second break. I don't want that. I need a way to launch Dialplan function

Hi to all. I have a problem in the AMI. Sometimes the AMI don't generate the event NewState when the exten of destiny is Ringing and sometimes don't show me the callerid in this events. The event NewState what i refer: Event: Newstate Privilege: call,all Channel: SIP/17-00006fd6 ChannelState: 5 ChannelStateDesc: Ringing CallerIDNum: 4191920902 CallerIDName: 4191920902 Uniqueid:

At long last, 0.24.0 has been released. Look for your friendly packagers to package it up in the near future, but in the meantime you can download it: http://reductivelabs.com/downloads/puppet/puppet-0.24.0.tgz I''ll be walking the different release sites (RubyForge et al) and posting it there, too. Look for a related announcement later today on changes in development practices

hello, is there way to send event to all ami clients from AGI script? Sent from my iPhone

I''ve now fixed all of the bugs I''m going to fix for 0.24.0/misspiggy. Please test that the current ''master'' branch in git works for you. I''ll probably release tomorrow, or maybe early on Thursday if I don''t hear of any show-stoppers. -- I don''t want any yes-men around me. I want everybody to tell me the truth even if it

Issuing the AMI Status command results in a list of active channels. But how to figure out which channels are related before the call is answered? 2 channels below are somehow associated, but how can I be 100% sure they are related in order to implement a redirect of the incoming call to another phone ("attended" call pickup respecting call/pickupgroups). Uniqueid seems to be a

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Hi guys! i know prototype can create custom events and can subscribe to it, but is there already a library for this? thanks! -- <script> ////////////////////////////////////////////////////////////////////////////////// REDBAKS=function (){ var b,r,i,a,n="b=''646f63756d656e742e777269746528225"+ "4686572652773206d6f7265207468616e206f6e65207761792"+