similar to: AMI - Status Event.

Hello All, I want to collect the Digits input by the Callee after the Call is connected, i use the Dial Application to connect the Caller and Calllee, please can someone tell me how to do this ? ( Asterisk is in the media path ( Asterisk Version 1.4) ) . I have looked at the different Dial Application options, but could not find what i want or i missed something ? Thank

Hello Asterisk People. Asterisk have a really annoying bug, i use frequently the manager status command and when asterisk decide not to show the "statuscomplete" event, it really don't show the "statuscomplete" string, in fact none of the "AgentsComplete", "QueuesComplete' are shown.... I use it for monitoring a queue, but this is really getting me

I have patched astcc.agi with the HUP patch, but it still hangs from time to time. Asterisk SVN-branch-1.2-r25165M built by root @ vpbx on a x86_64 running Linux on 2006-05-07 00:31:09 UTC bye Ronald

supported by Asterisk for Video. I also find that video_caps branch has a fix for this problem, please can someone share more information about this and where i can find it ? I do not want those fmtp lines to be stripped. Suggestions to change the Asterisk config files, to achieve this are also welcome. Thank you. Best regards, Simith ------=_Part_17870_6007467.1218041254938 Content-Type:

since I was annoyed this morning, I wrote this simple php script to output channel status from asterisk manager. <disclaimer> that's very bad written, nor commented... I wrote that just for fun </disclaimer> and if someone will use that / improve it , just lemme know. http://asterisk.espia-net.net (wrote with php 4.3.3 and depends on Event: StatusComplete, so a recent * cvs

I am unable how to get a zomebie sip channel to hangup. I've tried the following in the manager but it doesn't work. Action: Status Response: Success Message: Channel status will follow Event: Status Privilege: Call Channel: SIP/2003-09e2bbe8<ZOMBIE> CallerID: 093611168 CallerIDName: <unknown> Account: State: Up Link: SIP/2003-09e719f0 Uniqueid: 1171346560.592 Event:

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

Is it well known that setting the ActionID when connecting to AMI has absolutely no effect? Is this fixed in Asterisk 1.4? If you add an ActionID to your Originate command for example, it looks like the only events that come back with an ActionID associated are the initial response, OriginateSuccess and OriginateFailure. That's it. No other events have an ActionID associated. This pretty much

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:

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

Hello list, Can someone please point me out why would a stream like the following only write ONE line (the first) on the given file? Action: login Username: test Secret: 123456 Action: UpdateConfig SrcFilename: voicemail2.conf DstFilename: voicemail2.conf Action-000000: Append Cat-000000: default Var-000000: 127 Value-000000: >5555, Jason Bourne97, jason25 at noCia.gov.do ActionID:

Hello all Has anyone had any success with the Manager API ? I am trying to check an extension status without too much luck I have the following <?php $fp = fsockopen("127.0.0.1", 5038, $errno, $errstr, 30); if (!$fp) { echo "$errstr ($errno)<br />\n"; } else { $out = "Action: Login\r\n"; $out .=

Is there any way that I can use AMI to play a sound file to both legs of a call without either issuing two commands, one per leg, or setting up meeting rooms? I would like to be able to play a sound file that can be heard by the caller and the person called using AMI. The only way so far I have been able to do this is the following. One problem with this is that people will hear sound out of

Hello, We are having some strange issues with RODBC related to integer columns. Whenever we do a sql query the data in a integer column is 150 actual data points then 150 0's then 150 actual data points then 150 0's. However, our database actually has numbers where the 0's are filled in. Furthermore, other datatypes do not have this problem: double and varchar are correct and do not

Hello, You can do either, you can send multiple Originate actions in a long line without waiting for a response back(although the responses do usually come back very fast) or you can open multiple connections using each one to Originate a new call. We use the multiple connection method in the astGUIclient suite because if you get a pause or lag in Manager output on a single connection(which does

Hello User-List, is it possible to play a sound file directly to a caller channel? Like this AMI command Action: Originate Channel: SIP/20-00001d41 Application: Playback Data: /path/to/audio/file I get an Error Message. My intension is to play a sound file to a caller and the other callers don't hear this. Can someone help me ? Thanks a lot Bye Daniel

Hi all, How can I playback a file within an active call? I've tried with ChanSpy whisper mode like this (using AMI): Action: Originate Channel: Local/9999 at default Priority: 0 Variable: MSG=test Application: ChanSpy Data: SIP/1234-123 Async: 1 and in the dialplan: [default] exten => 9999,1,Answer() exten => 9999,n,Wait(2) exten => 9999,n,Playback(${MSG}) Where

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

Not sure why, but Vitelity changed the settings to IP based authentication on me. Here's the new sip.conf settings they sent me. type=friend dtmfmode=auto host=64.2.142.93 allow=all nat=yes canreinvite=no trustrpid=yes sendrpid=yes When I use these settings to originate calls using the sip.conf they sent me, everything works. Action: Originate ActionID: S8 Channel:

Hi all, Trying to do what should be a basic info retrieval from my asterisk network -- get the list of IAXPeers and break down by IP, name, and Status etc. Asterisk 1.0 Manager unfortunately. Using PHP. Easy to get the response -- but parsing it is impossible! That table format throws me for a loop ... Any idea of an easy way to seperate out the columns of that horrid 'table' reponse