similar to: AMI ActionID.... Doesn't work

Displaying 20 results from an estimated 1000 matches similar to: "AMI ActionID.... Doesn't work"

2005 Sep 13
2
actionID on manager events
Hello, all! I'm looking at the wiki page and info on the mailing list and I'm getting conflicting info... I am using the manager API from the telnet CLI and I am testing creating calls with it. I login with events: on and I can originate calls just fine. However, when I set ActionID on an Originate, I cannot see anywhere where that actionid carries into the Event output. But I found
2005 Sep 15
0
AW: ***SPAM*** actionID on manager events
hi, afaik, the action-id provided with the OriginateAction should only show up in the OriginateSuccess or OriginateFailure event. Intermediate events that are generated when the channels are create will NOT carry the action-id of the originate. The async flag tells asterisk to process originates in parallel, i.e. if you have two users originating calls and NO async flag set, the second originate
2007 Dec 10
0
diferents events between ast1.2 & ast1.4 ??
Hi all, I'm new in the list, and I have a problem upgrading from asterisk 1.2 to asterisk 1.4: There is a diference from asterisk1.2 to asterisk1.4 in AMI events. When I do a call to a queue (with the same extensions.conf dial plan) with ast1.2 and ast1.4, in ast1.2 apper 3 newcallerid event in ast1.4 apper only 2. It is normal? anyone knows it? what is the reason? I
2006 Nov 29
12
What's up with the Manager Interface?!?!
The Asterisk Manager Interface is driving me nuts. Whoever wrote it should be drawn and quartered. Sometimes the data comes back separated by \r\n, and sometimes it's separated by \n. The whole thing is completely inconsistent, and trying to write any kind of API for it is -GHASTLY- Doug.
2008 Feb 20
0
Strange NewCallerIDEvent after channel are linked
Hi all, just for learning purposes i made a little gui frontend that visualizes incoming and outgoing calls in realtime, using the events of asterisk. I experienced a strange behaviour for outgoing calls. The callerid for the *called* person got changed to one of my own numbers, after the channels git linked. After looking into the flow of events i saw that asterisk keeps sending an
2008 Oct 28
1
AMI - Status Event.
Hello All, I'am a new Asterisk user, and i have the following question. The following is the Status of all open channels from my Asterisk system, which was received through the Asterisk Manager Interface ((AMI)). ==================================================================== action: Status actionid: 65066874_3# Response: Success ActionID: 65066874_3# Message: Channel status will
2005 Mar 10
3
SetCallerID({$NEWCALLERID})
I am trying to SetCallerID to a variable I have defined. This obviously is wrong. It actually sets the caller ID to $NEWCALLERID. I have search through the examples on wiki but wasn't able to find something similar to see what I was doing wrong. Could someone tell me the correct way to SetCallerID to a defined variable? exten => 2125551212,5,SetCallerID({$NEWCALLERID}) exten =>
2007 Feb 05
0
Callfiles to Meetme Fails (was: RE: Using Local Channels with Originate)
I have Meetme conferencing compiled for Debian as per http://powerontech.com/freepbx-on-debian.htm . I drop a callfile in the outgoing dir, and it intitiates a call to a local extension as a channel, but the call seems to block on the Meetme() command. That extension completes the outgoing Dial(SIP) command to my phone, announcing that leg is the only member of the conference, and just waits. If I
2007 Mar 30
1
call file vs. originate
I'm having trouble getting the manager interface to behave properly; specifically the Originate event. If I create an originate event as below, the calling phone will auto-answer (as it's supposed to) but the receiving phone never rings. It will timeout at 20 seconds. Action: Originate Channel: Local/201@from-sip2 Context: from-sip Extension: 154 Priority: 1 CallerID: John Doe
2006 Jun 14
3
SIP, Microsoft RTC, and Originate problem
Skipped content of type multipart/alternative-------------- next part -------------- Reliably Transmitting (no NAT) to 111.111.111.50:16666: INVITE sip:111.111.111.50:16666 SIP/2.0 Via: SIP/2.0/UDP 111.111.111.8:5060;branch=z9hG4bK360502ae;rport From: "asterisk" <sip:asterisk@111.111.111.8>;tag=as348de10b To: <sip:111.111.111.50:16666> Contact:
2019 May 29
2
AMI not responding correctly
I am communicating with Asterisk 13.18.3 over the AMI and issue the command: ActionID: 11 Action: command Command: core show calls And the response I get is: Response: Follows Privilege: Command ActionID: 11 --END COMMAND- But where is the call data? What is going wrong on this system? I confirmed the AMI connection has full read/write permissions. Why is the call data
2023 Jul 02
1
Get channel variables via ARI/AMI
>> You use the AMI action Getvar[1] which allows channel variables and dialplan functions. >> [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+20+ManagerAction_Getvar I actually tried that, and although I get “success” I never get useful data. For example: action: Getvar actionid: act1 channel: PJSIP/Twilio-NA-W-2-In-00000025 Variable: channel(pjsip,call-id)
2023 Jul 03
1
Get channel variables via ARI/AMI
The uppercase command made a difference. I now get a call-id as show below. However, does the call-id look valid? The @0.0.0.0 seems strange. action: Getvar actionid: act1 channel: PJSIP/Twilio-NA-W-3-In-00000028 Variable: CHANNEL(pjsip,call-id) Response: Success ActionID: act1 Variable: CHANNEL(pjsip,call-id) Value: 4decf884e3ae74595906283a74f7154e at 0.0.0.0 As well,
2023 Jul 02
1
Get channel variables via ARI/AMI
On Sun, Jul 2, 2023 at 4:39 PM TTT <lists at telium.io> wrote: > >> You use the AMI action Getvar[1] which allows channel variables and > dialplan functions. > > >> [1] > https://wiki.asterisk.org/wiki/display/AST/Asterisk+20+ManagerAction_Getvar > > > > > I actually tried that, and although I get “success” I never get useful > data. For
2005 Jan 21
0
Help DIALSTATUS gives ANSWER when line is BUSY?
I'm running Asterisk CVS-v1-0-12/20/04. I'm using PHP with Manager API Here is the code: #################################################################### # Make call #################################################################### $socket = fsockopen($ask_db,"5038", $errno, $errstr, $timeout); if (!$socket) { echo "$errstr ($errno)<br /\n"; } else {
2006 Jun 13
1
calleridname.agi patch to only overwrite name if it is missing
I edited the calleridname.agi patch to only overwrite the name if it is missing. The asteridex option still overwrites the name since it is our master list for known numbers. -- Steven calleridname.agi.patch: --- C:\Documents and Settings\steveb\Desktop\calleridname.agi-orig Tue Jun 13 14:37:09 2006 +++ C:\Documents and Settings\steveb\Desktop\calleridname.agi Tue Jun 13 14:37:09 2006 @@ -16,6
2008 Dec 05
2
AMI interface problem
I installed version 1.6.0.3-rc1 and my AMI application stopped working. I reinstalled 1.6.0.1 and it worked again. I reinstalled 1.6.0.3-rc1 and it stopped. Looks like a problem in the software to me. Following the same steps using the same code for the AMI and conf files for * I get bad behavior in 1.6.0.3-rc1 and good behavior in 1.6.0.1. I have this action: Action: Originate Channel:
2004 Aug 04
1
Identifying which call an event belongs to
Hi, I guess I need some help with management interface. I would like to watch calls through the management interface, but I don't know how to identify which call an event belongs to or in other words how to associate a call and uniqueid field of event. Let's say I send the following manager command: action: originate channel: sip/12125551111@pbx1 callerid: 12125551111 MaxRetries: 1
2007 Nov 23
1
AMI Newstate Ringing events -- Inconsistent caller id ?
Hello list, I'm observing what I believe to be inconsistent behaviour regarding "Newstate" AMI events for the "Ringing" state. As such I come to you asking for experience or advice: am I wrong or should I file a bug ? I present you a short introduction which I feel is relevant; however, if you want to go straight to my technical question, please scroll
2005 Sep 15
1
Originate not understanding 2 vars in setvars
Hi, I'm currently trying to originate a call with 2 variables set. I tried doing it via manager API and call File and both failed, because the vars were not separated. I'm using Asterisk 1.2_beta1 on this machine Can anyone here verify wether this is a bug or just a stupid error on my part? This is the callfile I tried to use, after the manager way failed: Channel: