similar to: Help DIALSTATUS gives ANSWER when line is BUSY?

Hi All! Let me explain the problem. When using the Originate? command from the manager api, the dialstatus variable returns results? for whichever phone picks up first, and in this case it is the IAX/2? connection. It doesn't matter if Zap/G2/XXXXXXX is set as the channel,? or an extension either. What I am ultimately trying to do is get the? dialstatus of the Zap/X/XXXXXXX channel, i.e.,

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 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

Originally posted on asterisk-dev with no response for 5 days, so posting it to the wider audience now. Asterisk Release 1.6.1.1 Scenario:- 1. 2 SIP peers (Zoiper softphone, if it matters) registered as 901 and 902 2. Using AMI, 901 is Originated 3. When 901 answers, it is Redirected to an extension "exten => dial,1,Dial(SIP/902)" 4. 902 rings, then answers 5.

Hi Everyone, I am testing Asterisk 1.8 AMI events. The voipinfo page on AMI events is specific to 1.6. I am wondering if the developers cared to write about the new events that are spit out in Asterisk 1.8 somewhere on the web? I checked the tar ball for asterisk 1.8 and documentation doesn't include this event: *Event: Unlink* Privilege: call,all Channel1: SIP/9999-00000029 Channel2:

hello perl experts i am working with "ast-rad-acc.pl" from http://www.voip-info.org/tiki-index.php?page=PortaOne+Radius+auth i dont know why $cdr{'DNID'} and $cdr{'CALLERID'} under 'sub send_acc {' are empty. i m successfully connected with asterisk manager and when call i hangup my perl application is getting that all other thing are ok but i dont know why only

Hi, In my application I jump to different extensions For example: [begin] exten => s,1,Goto(starts,s,1) [start] exten => s,1,Play(welkom) ..... exten => h,1,Goto(end,s,1) [end] exten => s,1,Macro(end_call) exten => s,n, Hangup When I look at my CDR record I see three different CDR's in my record. Is there a way to use one CDR on every call, and not

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 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

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

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

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:

Dear community members, I'm happy to announce that we now have code that allows you to use your XMPP (Jabber) client like a softphone to place SIP or PSTN (or whatever channel Asterisk supports) calls. The XMPP clients that support Jingle that I and others have tested are : - Pidgin (Linux, Ubuntu 9.10), version 2.6.2 : OK - Empathy (Linux, Ubuntu 9.10), version 2.28.1.1 : OK - Psi (Windows

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, 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

I am sending this again since I haven't get it back for twelve hours: When I originate call to IAXComm, more or less one of tree calls fails for no aparent reason. Originating calls to SIP clients works as expected. Anybody has similar problems? Is it asterisk or client problem? Asterisk log: Jul 15 00:00:04 DEBUG[1179663]: manager.c:1018 process_message: Manager received command

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 > >

Hello Samba-bugs, My smtp server does not recognize samba-bugs@samba.org to I send it to samba@samba.org I have setup a Samba server on a Compaq Alpha 21164 system with 512M RAM and running Compaq Tru64 (Digital UNIX) OS. First I am using Samba 2.05a, and it's read speed or write speed is very slowly on some client PC, it's about 500KB/s. Some PC works fine but some not

Friends, a new dialplan application is now available for testing : http://svn.digium.com/view/asterisk/team/phsultan/jabberreceive/ The corresponding feature request is located here : http://bugs.digium.com/view.php?id=12569 What can you do with it? Well, a direct usage of this application is to make an easy to use GoogleTalk voice gateway out of Asterisk. Here is an example (assuming the

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