Displaying 20 results from an estimated 200 matches similar to: "Help DIALSTATUS gives ANSWER when line is BUSY?"
2005 Jan 21
0
Manager API on gives the DIALSTATUS of the first picked up channel?
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.,
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
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
2009 Jun 30
0
Redirect with ExtraChannel on Bridged call give AMI event with second channel name AsyncGoto/...<ZOMBIE>
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.
2011 Apr 08
1
Documentation for Asterisk AMI Events?
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:
2005 Jul 18
0
why $cdr{'CALLERID'} and $cdr{'DNID'} are empty in perl agi connected with asterisk manager
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
2008 Feb 04
1
one CDR instead of multiple CDR
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
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
2010 May 20
0
Early injecting Jack between call parties
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
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 Feb 01
0
manager api events (pri vs pstn)
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
2010 Nov 10
0
Problem with AMI
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:
2009 Nov 30
0
Asterisk and XMPP Jingle : testers needed
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
2013 Jan 18
2
A smart way to use "$" in data frame
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:
2010 Jun 09
0
AMI Queue information about incoming call's channel before link
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
2004 Jul 14
0
Originate to IAXComm problem once again
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
2020 Feb 07
0
[asterisk-dev] Need feedback on the use of AMI events generated by MESSAGE requests
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
> >
2000 Aug 10
0
Speed problem on Alpha system and solution
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
2008 Jun 11
1
Asterisk and XMPP (Jabber) : testing new application JabberReceive
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
2008 Jul 16
2
[PATCH] stopmachine: add stopmachine_timeout v2
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