similar to: Missing Manager Events/Actions: Hold, Reconnect, Conference

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

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

Asterisk 1.0.3 TDM400P/TE410P Using originate() call progress "Events" normal progression on completed call ================ Event: Newstate State: Ringing Event: NewState State: up ================ On "pri Zap" channels call progress events will wait @ "State:Ringing" until call FAILS via timeout if number dialed is disco'd, out of service, etc. and produce

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 {

Hello all, I have a data frame dataa: newdate newstate newid newbalance newaccounts 1 31DEC2001 AR 1 1170 61 2 31DEC2001 VA 2 4565 54 3 31DEC2001 WA 3 2726 35 4 31DEC2001 AR 3 2700 35 The following gives me the balance of state AR:

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

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:

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

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, I'm developing an application using AMI and I need to get information about incoming call _before_ queue member answers it. I'm using static members (queue is pretty simple) and don't want to use chan_agent (I think AgentCalled event will do what I'm looking for). Here is what I'm getting now: 1. Newchannel event for incoming call followed by Newstate and Join. All these

Thank you for useful feedbacks! Here is the updated version. Could you put this on top of your patches, Rusty? Thanks, H.Seto If stop_machine() invoked while one of onlined cpu is locked up by some reason, stop_machine cannot finish its work because the locked cpu cannot stop. This means all other healthy cpus will be blocked infinitely by one dead cpu. This patch allows stop_machine to

Thank you for useful feedbacks! Here is the updated version. Could you put this on top of your patches, Rusty? Thanks, H.Seto If stop_machine() invoked while one of onlined cpu is locked up by some reason, stop_machine cannot finish its work because the locked cpu cannot stop. This means all other healthy cpus will be blocked infinitely by one dead cpu. This patch allows stop_machine to

If stop_machine() invoked while one of onlined cpu is locked up by some reason, stop_machine cannot finish its work because the locked cpu cannot stop. This means all other healthy cpus will be blocked infinitely by one dead cpu. This patch allows stop_machine to return -EBUSY with some printk messages if any of stop_machine's threads cannot start running on its target cpu in time. You can

Hi everyone, My manager interface seems to be producing wrong CallerIDs when internal extensions call each other. Can anyone see anything wrong in the configuration snippets pasted below? The following instance has extension 101 call 103. The phone does show the right caller ID, but notice that the manager interface has the CallerID as the target number (103). Thanks a lot for your time.

If stop_machine() invoked while one of onlined cpu is locked up by some reason, stop_machine cannot finish its work because the locked cpu cannot stop. This means all other healthy cpus will be blocked infinitely by one dead cpu. This patch allows stop_machine to return -EBUSY with some printk messages if any of stop_machine's threads cannot start running on its target cpu in time. You can

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

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

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I have successfully set up a conference room on my asterisk server, I have been trying to make the 'M' for music on hold option work (when the first person enters the room they are told they are the first and then they are supposed to hear music on hold) but it didn't matter which way I wrote it this feature wouldn't work. Basically it wouldn't allow the conference to be

Hi, I have the following problem that when someone connects to my conference and is the only member music on hold is played just for one second or less and then stops: [Feb 1 10:38:46] -- Started music on hold, class 'default', on channel 'SIP/sip.touk.pl-0083dad0' [Feb 1 10:38:46] -- Stopped music on hold on SIP/sip.touk.pl-0083dad0 I use the following command to