Displaying 14 results from an estimated 14 matches similar to: "How to detect if the call is being answered by Voice Mail?"
2009 Aug 17
0
Call back DIALSTATUS is empty
Hi,
Here is my problem. I am trying to get the Status of the call if the user
picked up the phone or not. It is coming as empty. Please help.
Here is my extensions_additional.conf file code:
[multi-dir-callback]
include => multi-dir-callback-custom
exten => _X.,1,Answer
exten => _X.,n,Playback(beep)
exten =>
2007 Oct 15
1
Answering Machine Detection
I am having a bit of a problem getting AMD to work on a new server. On
my regular office server it works like a charm. I am running Asterisk
1.4.13, Zaptel 1.4.5.1 on both machines. Both servers run CentOS 5 and
I am using a SIP trunk to send out calls (the same one on both servers).
Here is the output of a call on my office server:
-- Attempting call on Local/0445540881644 at CC2 for
2007 Dec 02
2
Answering Machine Detection
If i use AMD() or the code below, now the problem is the fax machine/modem detection and answer machine detection get detected as the same. If i need to seperate the two how do i do that? For example, if i use AMD() to detect an answer machine by saying any greeting exceeding 2.5 seconds is a machine, how do i distinguish between a fax/modem and a long greeting?
---- dave cantera
2010 Feb 03
1
CDR / billsec / originate / local chan
Hi All,
I have been running a environment with asterisk 1.4.20.1 for some time
now with no issue but have recently added some extra functionality
(enabled call recording via MixMonitor) and ran into some deadlock
issues which seem to be well documented with earlier 1.4.x releases so
have decided to take the plunge and upgrade. I decided to start testing
with 1.6.2 but have run into a couple
2007 Aug 29
1
Re: using Ajax.PeriodicalUpdater in iPhone
it''s probably the iPhone.
I mean, would you *really* want your phone to be connected to the internet
all the time?
On 8/28/07, richieright-Re5JQEeQqe8AvxtiuMwx3w@public.gmane.org <richieright-Re5JQEeQqe8AvxtiuMwx3w@public.gmane.org> wrote:
>
>
> Hi,
>
> I met a strange problems with PeriodicalUpdater in iPhone. This timer
> event will last only four or five
2009 Aug 31
4
How to stop IVR once system receives DTMF?
Hi,
We are trying to implement a complex business logic in Asterisk. Executing
"Wait_For_Digit" command after playing IVR. We want to stop the IVR once we
receive the digit. It is not recognizing the Digit until it completes the
IVR. How can we stop the IVR once we receive the digit?
Thanks
BB
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2006 Mar 26
0
UK EI
I'm using a Digium TE411P connected to a UK switch (EuroISDN).
Everything is working, but if I dial a busy number (from SIP) is seems
to stay busy until I hang up, even though the dial-plan drops through
some other stuff using CALLSTATUS variable (i.e. S-BUSY), none of the
timeouts come into play.
Any ideas?
Steve
--
NetTek Ltd UK mob +44-(0)7775 755503
UK +44-(0)20 79932612 / US
2010 Feb 08
0
originate, local channel and failure extension
Hi All,
I am in the process of migrating from 1.4.20 to 1.6.2.x and have
stumbled across a number of "differences" between the 2 versions that
are forcing me to use local channels in my dialplan (mainly centered
around the different behavior of CDR fields in the 2 versions) .
Previously, I would place a call via an AMI Originate action similar to:
action:.Originate..
2006 May 23
3
AGI ?
Hi All,
I have been attempting to get an AGI LCRdialout script to work.
Basically what I need to have happen is when someone dials out a number
the script check to see if it is local if so, go out the ZAP channel. If
the ZAP channel is busy, go out the IAX channels, if IAX is all busy, go
out the SIP channels. Here is a sample of what I have in my script.
#!/usr/bin/perl
use strict;
use
2009 Nov 16
1
MixMonitor and Call Latency during conversation
Hi,
We are using MixMonitor to record the call. When the call is bridged, the
latency is significant. We tried to increase the internet speed and the
server RAM and processor speed and still we are having that issue.
We use VoiceTrading and Gafachi's Termination minutes to make calls. As we
are in US and VoiceTrading in Europe, somebody suggested to move the
termination minute provider
2010 Nov 24
1
Disable connected line updates for dahdi PRI channel
Hi,
Starting in Asterisk 1.8.0, Asterisk supports connected line updates.
This is fantastic for SIP. How can I prevent them from being sent to a
PRI channel?
I'm having problems when a call is answered by an internal SIP
extension, then transferred (blind or attended) to another internal SIP
extension. One of my PRI providers can't handle the ROSE_ETSI_EctInform
APDU and drops the
2010 May 12
3
Asterisk core dumping on SendFax with FFA
Hi All,
I seem to have stumbled on a bit of a problem. When trying to send a fax
with Fax For Asterisk on 1.6.2.x (have tried 1.6.2.5, 1.6.2.7 and the
current svn version, with FFA 1.2 I get a core dump each time.
Here is an extract form the console:
[May 12 22:47:09] DEBUG[22584]: app_queue.c:1084 handle_statechange:
Device 'SIP/vltb-sbc01' changed to state '1' (Not in use)
2006 Jun 27
2
Callstatus on bridge IAX2 <-> ZAPTEL is always "answer" even if the call fails
Hello,
In my asterisk box, i have a zaptel card connected to my analogic pstn line.
I'm using a IAX2 client to call outside :
IAX2 client <--> Asterisk <--> Zaptel card <----> France telecom line
When checking cdr logs file, i always have an "ANSWER" on call status when
call on this trunk, even if the final destination does not answer. Is
"ANSWER" the
2006 Apr 05
15
How to restrict simultaneous phone registrations
Hello all,
I am looking for a way to restrict users from logging in two separate
phones with the same authorization name/password at the same time.
Meaning, I only want users to be able to place a call from one phone in
one location, but have the ability to move from computer to computer.
Has anyone found any sort of solution for this type scenario?
Thanks,
Bryan Mahin
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