similar to: AMI Queue information about incoming call's channel before link

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

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 {

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

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

This may be better off on the developer list, but I thought I would see if I was way off-base before I went there. I am working on a manager CTI client (currently for windows but with hopes of porting it elsewhere later). [Hold/Reconnect] I have many of the features working. I can originate calls (using the call-in/call-out originate that the manager provides) display call-related information

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 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:

It just dawned on me as I was playing with the manager interface - it can't be very difficult at all to write an Win32 app that serves as a "lamp field". Between 'Newchannel', 'Newstate', and 'Hangup' events, all of the information is there. I've heard several requests for line appearances, but mgcp and sccp channels don't currently include support.

I am originating calls through the Manager Originate API command. I can track failures (through the OriginateResponse event) I can track answered calls through the OriginateResponse event) There may be occasions where I need to cancel some outbound calls whilst they are ringing. Here's my problem: How do I know what the channels are in order to cancel them ? I can get a

I'm creating an AMI client and I only want to get newchannel events (as well as responses to any actions I initiate). What would I set the eventmask to to only get the newchannel events? For anyone else looking...is there a table somewhere online that maps events to their eventmask categories? I checked the asterisk wiki and voip-info but can't find this... -------------- next part

Hi, I'm writing a Java client/server application that talks to the Asterisk manager interface via the asterisk-java stuff. The idea being it will give you an app to run on your desktop that monitors your phone essentially. Once I've got something vaguely working it will be released under the GPL and hopefully people will contribute to it etc... As part of this, I'm currently

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. My configuration is Elastix 1.5.2-2 (asterisk 1.4.24, libpri-1.4.3-5, dahdi-2.1.0.4-7 ) and OpenVox d210e connected to telco provider (Euro ISDN). Here is my /etc/dahdi/system.conf: # Span 1: TE2/0/1 "T2XXP (PCI) Card 0 Span 1" (MASTER) HDB3/CCS ClockSource span=1,1,0,ccs,hdb3 # termtype: te bchan=1-15,17-31 dchan=16 echocanceller=oslec,1-15,17-31 # Span 2: TE2/0/2 "T2XXP

From: Haiyang Zhang <haiyangz at microsoft.com> Rename camel case variables in channel.c Signed-off-by: Haiyang Zhang <haiyangz at microsoft.com> Signed-off-by: Hank Janssen <hjanssen at microsoft.com> --- drivers/staging/hv/channel.c | 729 +++++++++++++++++++++--------------------- 1 files changed, 368 insertions(+), 361 deletions(-) diff --git

From: Haiyang Zhang <haiyangz at microsoft.com> Rename camel case variables in channel.c Signed-off-by: Haiyang Zhang <haiyangz at microsoft.com> Signed-off-by: Hank Janssen <hjanssen at microsoft.com> --- drivers/staging/hv/channel.c | 729 +++++++++++++++++++++--------------------- 1 files changed, 368 insertions(+), 361 deletions(-) diff --git

Hello. I've been trying to use the AMI to originate phone calls. I'm trying to call the SIP phone 'zoiper' with the SIP phone 'yziquel'. So, the AMI interaction is: > Action: originate > Channel: SIP/zoiper > Exten: yziquel > Priority: 1 > Timeout: 30 > Context: internal > > Response: Error > Message: Originate failed > > Event:

Hello, i have a problem with the hangup cause received from the AMI in the Hangup events. All calls that arent answered after ringing are returning hangup cause 16 (normal clearing) instead 19. Im running asterisk 1.4.11, the calls are generated to a SIP peer using the AMI originate command. This is the 'sip debug' output: Reliably Transmitting (no NAT) to 192.168.0.70:5060: INVITE sip:1

On Thu, Feb 6, 2020 at 12:34 PM sduthil at wazo.io <sduthil at wazo.io> wrote: > On 1/29/20 2:31 PM, George Joseph wrote: > > For those of you who actually process SIP MESSAGE requests... Do you > > use any of the AMI events generated by the "Message/ast_msg_queue" > > channel? We want to change that channel to an "internal" channel that > >

From: Haiyang Zhang <haiyangz at microsoft.com> Remove camel cases from vmbus channel functions Converted the function names, local variables to lower cases. More patches will be submitted soon to remove the remaining camel cases. Signed-off-by: Haiyang Zhang <haiyangz at microsoft.com> Signed-off-by: Hank Janssen <hjanssen at microsoft.com> --- drivers/staging/hv/channel.c

From: Haiyang Zhang <haiyangz at microsoft.com> Remove camel cases from vmbus channel functions Converted the function names, local variables to lower cases. More patches will be submitted soon to remove the remaining camel cases. Signed-off-by: Haiyang Zhang <haiyangz at microsoft.com> Signed-off-by: Hank Janssen <hjanssen at microsoft.com> --- drivers/staging/hv/channel.c