Displaying 20 results from an estimated 7000 matches similar to: "Manager cmd: originate without picking up thefone?!"
2006 Feb 13
1
Manager cmd: originate without picking up the fone?!
Hi There,
we are developing a dialer application using the java lib
to interface with the asterisk manager protocol. It works
fine so far. The only problem we have is that if we use
the "originate" command the user is required to pick up
the fone _bevore_ asterisk will originate the call to
the desired destination.
What we would like to do is to place the call, check if
the other end
2006 Jan 30
1
Manage api- Matching 'Newchannel' event with the 'Originate' command
Hi all,
When the 'Originate' command is issued with 'Async' open set to 'yes', I got the response right away with the correct 'ActionID'. What follows is the 'Newchannel' event with a 'Channel' ID, but their is no 'ActionID' to tie it back to the command. How do you guys deal with this?
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2006 Mar 13
1
Need help implementing call center featuresofAsterisk
It sounds like Naren and company has their own CRM application. They need a predictive dialer that allows third party app integration.
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com]On Behalf Of Matt
Florell
Sent: Monday, March 13, 2006 8:03 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
2005 Jul 25
4
Voicemail and musiconhold sound stopped working
Hi,
i am testing stuff for a couple of days now with Asterisk CVS-v1-0-07
and everything worked fine sofar when suddenly the voicemail and
musiconhold sound output stopped working.
The voicemailmenu still works though. I can see the voiceprompts etc
in the debug messages on the asterisk CLI but i cant hear
anything. Everything else works fine though. I can call out
fine etc. I did some network
2004 Jan 09
1
Asteriks as SIP<>H323 Proxy?
Hi,
is it possible to use Asteriks for translating SIP to H323 and vice versa?
I am looking to implement the following Setup
SIP UAC <-> SIP-Server <-> SIP/H323 Proxy <-> H323 Server <-> H323 UAC
Basicly i want SIP fones to talk to H323 fones and and SIP Fones to
access PSTN Gateway(s) in a H323 network.
Anyone got something similiar running? Any ideas?
best regards,
2008 Nov 14
0
Originate on AMI
Hello all,
I'm trying to develop a dialer interface from my
application, basically to originate calls on asterisk using the Manager
Interface. During this development I came across a situation and I realized
that the asterisk Originate command could be a little better than it is
actually.
Anyone who tried Originate for a "commercial" use will
2007 Sep 10
5
Asterisk Manager API - Originate command
Hi all,
Just ran into some issue with the originate AMI command. It seems that
there is a limit of around 120 calls I can place with the originate
command simutanously. By that I mean sending Asterisk a lot of originate
command very fast. Anyone know if there is a limitation? Thnx.
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2006 Feb 08
6
Connecting to live calls
Hi all,
Is there a way to connect two live calls through the manager api without directing them to a meeting room? Currently, I can connect them by sending them to a meeting room. However, I don't know what the overhead is, and I kind of think that if I can connect them or link them up, the overhead would be minimum.
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2006 Apr 27
12
PRIs from two different telco
My TE411p does not seem to like to have two PRIs from different telcos
(span 1 and span 2). I can get one working, but not the other. However,
if I use vpmsupport=0 when loading the wct4xxp module, they both work.
But here is the problem, vpmsupport=0 disables the on board echo
cancellation. Any ideas?
BTW, here is zaptel.conf
span=1,1,0,esf,b8zs
span=2,2,0,esf,b8zs
bchan=1-23
dchan=24
2005 Feb 28
1
Manager "Message: Originate failed" beinggenerated when callee does not pick up
<<I am getting "Message: Originate failed" even the phone is ringing on the other end of the line.>>
Originate will ring your own extension first and when you pick up, call the other number. If you don't pick up your extension, you will receive the message you see.
Bill Seddon
________________________________
From: asterisk-users-bounces@lists.digium.com on behalf
2003 Jun 18
2
Forward stepwise procedure w/ stepAIC
I'm attempting to select a model using stepAIC. I want to use a forward
selection procedure. I have specified a "scope" option, but must not be
understanding how this works. My results indicate that the procedure begins
and ends with the "full" model (i.e., all 17 independent variables)...not
what I expected. Could someone please point out what I'm not
2005 Jul 18
2
Comments on Areski Calling Card Solution plz
Hi,
can anyone who has the Areski Calling Card solution on Asterisk
working comment on it? Is is stable enough for a production system?
Any pros and cons?
thx,
Arnd
2006 Apr 03
3
Monitor or mixmonitor
Hi all,
I am setting up a script to record all the call. There are two app for recording. "Monitor" and "Mixmonitor", one mixing the audio on the fly and one mixing it at the end but also allow a option not to mixing the audio at all. If mixing the audio on the fly is not that taxing on the CPU, I would like to use 'mixmonitor' app. My question is, what is penalty on
2018 Feb 15
2
OpenShift Origin Install
Hi,
I'm trying to install OpenShift Origin on a CentOS 7 host (just for
initial testing), and I'm trying to follow the instructions from here:
https://wiki.centos.org/SpecialInterestGroup/PaaS/OpenShift-Quickstart
On that page we need to run: "atomic-openshift-installer install" to
configure OpenShift... after run the script it throws this errors:
Failure summary:
2006 Nov 14
0
Redirecting Calls
Hello All.
I am stumped, please help me out..
I have the following setup:
VOIP provider = VOIP GW (asterisk GW1) = VOIP server (asterisk - VS1)
The gateway is there to get around the limitations running on the VOIP
server.
I can call out from and receive calls VS1 no problems at all. However,
when I try
and redirect an inbound call out via the GW, it drops out.
I have found that if I
2006 Mar 22
6
Can this box handle 8 T1s (PSTN) with Asterisk?
Hi all,
I am handed a project to setup *. The requirement is that it can handle
8 T1s. Half of the calls coming into the system will be routed to SIP
extensions (with transcoding). The machine we have in our disposal is a
new dual Xeon 3.2gHz server with 2g of ram and an dual 1000mb nic. Voice
will be coming in from the PSTN (through 2 quad digium cards) in
g711ulaw, and most of the time will
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.,
2004 Jan 09
4
Erro C0000001
Hi everybody
I get it the follow error:
The System can not log on.
Between parentheses (C0000001).
What the means of this, please?
--
Gilberto Nunes
Suporte Rede Bonja - Bom Jesus/Ielusc
Fone: 433-0155 - ramal 235
www.ielusc.br - suporte@ielusc.br
Linux User n? 199930
ICQ #136176504
--
Gilberto Nunes
Suporte Rede Bonja - Bom Jesus/Ielusc
Fone: 433-0155 - ramal 235
www.ielusc.br -
2009 Sep 02
1
AMI Originate Commands executed in sequential Order problem
Hi,
I noticed that asterisk manager interface will only accept the originate
commands in sequential order. For example, if I want to ring two extensions
through the AMI, and while first extension is ringing, AMI won't execute and
ring second extension until first extension has answered the call.
Anybody has any ideas as I had the same results even tested with telnet
commands to AMI interface.
2007 Dec 14
0
Problems booting DomU on CentOS 5.1
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Hi folks...<br>
<br>
I''m having problems to boot an DomU in CentOS5.1. I already tried
booting up several distrubutions and all of them boots till they reach
the rc.local. When it reachs