similar to: Transfer cmd via AsyncAGI

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

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

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 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

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]

Hi all, I have an external application commanding asterisk by AMI and AsyncAGI. I also have a dialplan like this: ; AsyncAGI extensions exten => _8.,1,Noop(entering in AGI loop at 8 ${EXTEN}); exten => _8.,n,AGI(agi:async); exten => _8.,n,Hangup(); ; Meetme extensions exten => _1.,1,Noop(Conference ${EXTEN} ${CONTEXT}); exten =>

Hi Is there any way to detect inactivity on channel when AsyncAGI is used? I want to detect whether application handling calls using AMI & AGI has stopped responding. Alternatively, how can dialplan check if there is any AMI user connected and decide dial plan execution? Thanks & Regards, Amit Patkar -------------- next part -------------- An HTML attachment was scrubbed... URL:

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,

Is it possible to simply send a message to appear as an AMI message/event, from the dialplan? For example exten =>123,1,ami(myEvent, param1, param2) and in the AMI a message appears like: Event: myEvent Privilege: call,all Channel: PJSIP/misspiggy-00000001 Uniqueid: 1368479157.3 ChannelState: 3 ChannelStateDesc: Up CallerIDNum: 657-5309 CallerIDName: Miss Piggy

Hi I am using following say.conf file. Its a default file, which comes with Asterisk installation. When I call SAY DATETIME AGI function, it simply returns without playing date & time. Where as if I use mode=old setting, it works. Is this a bug or mode=new is not implemented for SAY DATETIME AGI function? [general] mode=new ; method for playing numbers and dates ;

It means, AMI application is no more running or crashed or lost network connection with asterisk server. In such cases call is neither answered nor disconnected by Asterisk. I want to detect such state and jump to next dial plan to answer or reject the calls Regards Amit Patkar On September 20, 2016 8:07:23 PM GMT+05:30, Matthew Jordan <mjordan at digium.com> wrote: >On Sat, Sep 17,

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

I'm using a Snom 320 in a CAP position and the receptionist wants to do blind transfers. OK, no problem so far. Now she has asked me how to UN-transfer a call, as in, she transfers a call and wants to hook the call back before it connects (she wanted to tell the caller additional information for example) I don't think that this is possible as once my dialplan starts using Dial()

I am implemented a simple project management application. Each project has various events, and each event can be a different type. Some event information is common, such as name, start date, close date, and comments. I have a projects table which has_many events. My plan is to have several sub-event tables, like event_get_access which will contain an event_id field to link it to table events as

I am having trouble understanding how to use this. I want to transfer certain incoming calls from an IAX ITSP based on caller ID. From what I can make of the docs, I thought I need to do something like this... exten => _NXXNXXXXXX,n(nocid),transfer(1000) exten => _NXXNXXXXXX,n,noop(boo,${TRANSFERSTATUS}) exten => _NXXNXXXXXX,n,hangup exten =>

I've been trying to get an AGI "transfer" to work for several weeks now. It isn't error-ing out, but it isn't working either. I can't use "dial" in this case due to what I'm trying to accomplish. Does an AGI Transfer actually work? -= Info about application 'Transfer' =- [Synopsis] Transfer caller to remote extension [Description]

Some background... We use AMI and AsyncAGI to be able to receive events about calls (and other Asterisk details) and control it from our application. Works great and have about 100 sites (some newer, some older) without issues. I was notified this morning about a customer who had something strange happen and I can't explain it. Asterisk 16.23.0 and PJSIP. Call comes into Asterisk. Asterisk

Hello List! We're having trouble making call deflection on ISDN PRI. We would like to transfer a call to an external extension but keeping the callerid of the caller so it can be presented to the receiver of the transferred call. At the time we're using Zaptel 1.4.5.1, Asterisk 1.4.11 and Digium hardware TE420B. We've ordered the service (CD) from the phone company. The

Hi, I'm bringing this discussion here from http://www.moythreads.com/wordpress/2007/12/24/asterisk-asynchronous-agi/ about how to manage stopping a playback on a extension previously launched with AsyncAGI and redirecting the call to another exension. If I make the Redirect without a playback, the Redirect works: http://docs.google.com/Doc?id=ahfnfrcrh3rr_30f7fzq4hd But if I make the

Hello, as a variation of our issues with Adhearsion calls dropping when an INVITE comes in for a bridged call, I now have a new issue to contend with. Our call is in an AsyncAGI application, and has been bridged to another channel. The provider that supplies the DID sends a polling reINVITE every 15 minutes (it's a documented Metaswitch behavior amongst others). The reINVITE is seen as a new