On Thu, 17 Mar 2005 15:18:02 +0800, Ronald Wiplinger [EMAIL PROTECTED] wrote:
I tried wiki, but I got too many pages (I think all of them), ...as answer.
I want to write an agi.
I need a HOW-TO, is there anything available?
It depends. What language do you want to use?
Chris
On Tue, 29 Mar 2005 09:53:03 +1000, Rod Bacon
[EMAIL PROTECTED] wrote:
What I would like to know is has anyone found an open-source billing
platform that performs basic billing functionality (pre/post) from
RADIUS and/or Asterisk CDR and is written (well-written) in either PHP
or PERL.
What
On 5/19/05, Michael Stahl [EMAIL PROTECTED] wrote:
May 19 13:48:51 WARNING[7860]: Not a wav file 49
May 19 13:48:51 WARNING[7860]: Unable to open fd on
/cygdrive/e/pbx/voicemail/default/2460/INBOX/msg.wav
I'm seeing the same thing. I'm running -HEAD, checked out earlier
this afternoon,
On 5/31/05, Christopher Snell [EMAIL PROTECTED] wrote:
May 31 18:35:27 WARNING[21838]: format_wav.c:135 check_header: Not a wav file
49
- In my voicemail.conf, I have 'format=wav49' because the default
settting of 'format=wav49|gsm|wav' yielded voicemail files that were
nothing but loud
Google and voip-info.org will have answers to all of your questions.On 5/17/06, Jonathan Gonzalez
[EMAIL PROTECTED] wrote:Hi group,i'm brand new and i would like to ask about soekris hardware. I read
along the web but i have some doubts that i think can be solved here.My question are the
For redundancy on the PRI side, we plug our PRIs into Redfone
Networks' foneBRIDGE boxes. They've worked quite well for us. As
others have stated, use short registration periods combined with some
HA software to handle your SIP redundancy. You might also look into
load-balancing SER proxies.
Hi,
We're using Polycom IP501 SIP phones (app version 1.6.4.0043) with
Asterisk 1.2.9.1. I set up call parking last week and for a while, it
worked great. It stopped working yesterday, all of the sudden. What
happens is that when the phone user dials #999 (our parkext), the call
does not get
Hi,
I posted earlier about Call parking breaks suddenly. I believe that
I have narrowed this down to a problem with DTMF detection and the
Sangoma A101 card that we use.
Earlier, DTMF detection was not working at all. Then, I set
'relaxdtmf=yes' in zapata.conf and it works...sort of. When I
On 7/12/06, El Flynn [EMAIL PROTECTED] wrote:
Are you only having this problem for call parking? Any issues when the caller is
navigating an IVR?
We're not running an IVR on this particular system. Here's the
strange thing: the DTMF is not coming from the inbound caller but
rather, the
Alex,
Did you ever get an answer to this or figure it out? I'm having the
exact same problem.
In fact, I can run the ast-playlinein and pipe it to aplay and hear
the sound over the sound card just fine. I have a suspicion that
Asterisk does not like the format that arecord is spitting out or
Hi,
I've followed the instructions on the Wiki for pulling music-on-hold
from my sound card's line input. It doesn't work, however. MoH
starts and immediately stops. Apparently, I'm not the only person
having this problem. I'm thinking that maybe arecord(1) is not
sending the right kind of
For our stores, it would be nicer to have some kind of device that
automatically mutes our music before playing input from the Asterisk
pager. We already have a store full of speakers, no reason to
duplicate them. Has anybody heard of such a thing?
On 7/21/06, [EMAIL PROTECTED] [EMAIL
Sorry for the old thread revival...I bought three Sangoma A104 cards
to use as T1 (not PRI) data cards in an OpenBSD router. I was
disappointed to find out that trunking is not supported with this
configuration. I contacted Sangoma and was told that they would look
into it but I haven't heard
Hi,
How do you folks handle really large queues (350+ simultaneous
callers) in your Asterisk PBXes?
We're going to be bringing in around 16 PRIs' worth of inbound
callers, doing skills-based routing, and queuing them up for
approximately 200 agents.
What's the best way to handle all of these
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