Displaying 20 results from an estimated 1000 matches similar to: "G729 on wrong bus"
2006 Nov 20
2
Recording g729
<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
<html>
<head>
<meta content="text/html;charset=UTF-8" http-equiv="Content-Type">
<title></title>
</head>
<body bgcolor="#ffffff" text="#000000">
<font face="Helvetica, Arial, sans-serif">Before ordering I want to be
2006 Nov 17
5
spc.exe
Does anyone have a copy of spc.exe they could send me?
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061117/ac0b6a44/attachment.htm
2007 Nov 28
4
G729/MOH Quality
Does anyone have any opinions on the music on hold quality over G729?
The stock files seem to sound terrible over it, this is enhanced further
by calls coming from the PSTN via a Zaptel gateway. I am only using the
stock wav files and have not attempted to use much else so far.
I've ruled out timing issues on the system generating the MOH itself
(ztdummy on the PBX itself, our Zaptel gateway
2011 Mar 24
4
Issues with Digum Repos / AsteriskNOW & Bad Packages
I wish to use AsteriskNOW (the Digium repository + CentOS) with imap
voicemail storage and Asterisk 1.4.
After having installed AsteriskNOW with Asterisk 1.4 and Asterisk GUI
I run the yum package manager and replace voicemail with imap
voicemail and attempt to start Asterisk, however the voicemail module
is not loaded:
[Mar 23 23:30:03] WARNING[12592]: loader.c:382 load_dynamic_module:
Error
2010 May 29
2
Switchvox vs Asterisk codebase
Does anyone know what version of Asterisk Switchvox uses, and if it is
modified in any way? FWIW, I am dealing with a provider that claims
compatibility with Switchvox but not Asterisk for their SIP trunking
service.
2008 Jan 14
2
G.729 pre-compiled binaries and Asterisk 1.2.x.
Asterisk 1.2.24 seems to crash repeatedly under any substantial call load
(and sometimes without a substantial call load - just one SIP leg is
enough to do it) when using the G.729 pre-compiled binaries from:
http://asterisk.hosting.lv/
As per:
http://www.voip-info.org/wiki-Asterisk+G.729+Licensing
Time to crash is variable, but seems to require at least an hour of
production performance
2007 Jul 18
1
Any way to determine remote Asterisk version
A long time ago (Asterisk 0.x, 1.0.x) my experience is that there were alot
of interoperability issues, a common troubleshooting issue was to make sure
all endpoints where using the latest version of Asterisk. I have not seen
these issues in a while.
However I've been working with a customer of mine and this ITSP called IP
Communications (IPComms.net) well turns out we have had constant
2007 Jan 16
4
Audiocodes GPL
I have some Audiocodes units which appear to be running Linux,
according to the unit's own "System Log"
kern.warn Linux version 2.4.21openrg-rmk1 #2 Wed Aug 30 17:05:29 IDT 2006
However my contact at Audiocodes claims otherwise
On 12/4/06, Yaniv Nizan <Yaniv.Nizan@audiocodes.com> wrote:
>
>
>
> I doubt that we are running Linux on the MP-202. Perhaps there is a
2006 Oct 22
3
Audiocodes MP-20x
Has anyone used the AudioCodes MP-20x?
http://audiocodes.com/Objects/Analog_Telephone_Adapter_Series_MP_20X.pdf
Seems like a good device, but I can't seem to find anyone actually using
them...
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061022/6ca85b8c/attachment.htm
2007 Feb 05
5
Asterisk Faxing Support
Asterisk 1.2 has no support of t.38 whatsoever, the call will drop
before t.38 is ever utilised, not even pass-thru.
1.4 Adds support for T.38 pass through only and no other sort of
faxing, the endpoint must support T.38 and you must send your call to
a T.38 gateway and you must not use NAT anywhere in your network and
you must enable re-invites which could cause CDRs not to reflect the
true
2003 Oct 03
9
No Ringback on Iconnect
When I place a call using Iconnecthere as my sip provider, I hear no
ringback when making a call. Does anyone else have this problem or
offer any suggestions? Thanks, Kevin
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031003/b048f72f/attachment.htm
2004 Jul 07
4
VoicePulse Connect DID Problems
I have a DID with VoicePulse Connect, but the sound quality is horrible, it is
often choppy and the caller's voice cuts out for 2-3 seconds at least once a
minute, I have contacted VoicePulse many times, and they do not do anything
about it! Does anyone have any similar problems? It isnt my Asterisk config
because I have 0 problems using NuFone.
2003 Aug 07
3
SIP Lines
Instead of using a PCI card is it possible to use an outside SIP service
for "CO" lines?
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030808/33b131e3/attachment.htm
2007 Nov 30
3
Only call me once
Anyone have an idea how to implement a phone number that can only be
called once? The first time it will process normally and any
subsequent calls will be rejected.
2008 Jul 25
2
openSUSE Asterisk Packages
Does anyone know who maintains the asterisk packages in the openSUSE
buildservice? They are not updating Zaptel with their kernel updates
and I want to get that matter corrected.
I submitted to them a bug report but they seem to not care...
https://bugzilla.novell.com/show_bug.cgi?id=407408 ... usually within
24 hours a bugreport is assigned or some sort of comment is made.
2008 Nov 19
4
question about connecting with Mobile Base Station
Hi,
Is it possible to connect Asterisk with a mobile base station to handle call
switching? What kind of protocol will I need to use to convert to sip?
Any pointer or info will be greatly appreciated.
Best Regards,
Mark
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081119/e74ef6b1/attachment.htm
2010 Dec 04
3
Polycom Park by EFK
Has anyone gotten one-touch call parking to work on Polycom phones via
the Enhanced Feature Keys feature working? I've looked at various
examples, they appear correct, but the phones (501, 3.1.x firmware)
show the Park button while in a call but this does not actually cause
the call to be parked. Doing a SIP debug, I don't see that anything is
transmitted as a result of pressing the call
2006 Oct 31
7
Asterisk Call Statistics
Hi Folks,
I would like to recover all information about the calls, incoming
calls, call time, call history, etc in a Web Format, are there some
open source aplication for Asterisk that be easier for use. Pls
anything suggestion will be very appreciate.
Thanks
Rgds.
--
Omar E.P.T
-----------------
Certified Networking Professionals make better Connections!
http://omarept.blogspot.com/
2006 Oct 25
1
WiFi Phones (was Looking for Wireless Heaset for Polycom 501)
Martin:
I had seen your other post and sent you a message off-list, but I never got
a response. What do you feel is the most lacking that does not make it ready
for a production enviroment.
-
I've been using a SIP deskphone in my office and usually some sort of ATA at
my house, both as the primary phone. I've also had mobile phones from almost
every carrier. Each one of these devices
2008 Oct 01
1
Software patents (was G723 on asterisk 1.4.1)
On Wed, Oct 1, 2008 at 6:34 AM, Andrew Joakimsen <joakimsen at gmail.com> wrote:
> On Sun, Mar 23, 2008 at 11:34 PM, Tilghman Lesher
> <tilghman at mail.jeffandtilghman.com> wrote:
>> It is completely illegal in any country that recognizes patents.
>
> You mean countries that recognize software patents, right?
As resident of country where the file is hosted - yes we