On Thu, 2011-09-01 at 21:32 +0530, RSCL Mumbai wrote:
>
>
> My main interest of being on Virtual platform is portability / Backup.
> In case of any h/w issues, or crashes, simply copy the VM on to
> another box and you are up in minutes.
>
>
> Sanjay
> --
Doing that right now, although
Hi all,
Last couple of days i've arguing with my colleges about presence-info.
>From the asterisk-bible and the wiki's i learned that it is possible to
let asterisk do some of the presense-info by means of the jabber.conf
file and a seperate xmpp-server.
On the other hand, most soft-phones are c
On Wed, Aug 31, 2011 at 11:55:37AM -0400, Andrew Latham wrote:
> On Wed, Aug 31, 2011 at 11:49 AM, Carlos Chavez
> wrote:
> > only thing to keep in mind is to always connect the units in a
> > predetermined order to the USB ports so you do not mess up your
> > configuration.
> I am sure that Tz
I have already checked that. The voice mail was disabled when it first
occurred. So I set the voice mail up and it still happens but with no
new messages.
On 9/1/2011 3:33 PM, Carlos Chavez wrote:
Voicemail indication on the FXS port? I you have voicemail configured
the ring is indica
Voicemail indication on the FXS port? I you have voicemail configured
the ring is indicating that the extension has a message waiting.
On Thu, 2011-09-01 at 14:51 -0500, Chris Ramirez wrote:
> I have recently setup Trixbox 2.6.1 on a machine and configured it
> with an FXO and FXS module.
I have recently setup Trixbox 2.6.1 on a machine and configured it with
an FXO and FXS module. I can make and receive calls just fine so there
is no problem with the configuration of how the ports are set. The
problem I am having is when I miss a call. The phone will ring 15
minutes later and c
On 11-09-01 10:35 AM, Tim Nelson wrote:
- Original Message -
On 11-09-01 07:04 AM, Tim King wrote:
I have found numerous claims that 1.8 can do T.38 gateway with a
patch,
however I am yet to find the patch or any instructions on
implementing it.
Anyone have a link?
Asterisk-10.0.0-bet
On 11-09-01 03:30 PM, Tzafrir Cohen wrote:
On Thu, Sep 01, 2011 at 10:32:52AM -0400, Jeff LaCoursiere wrote:
I tried and failed with VirtualBox too. Timing seemed impossible to
maintain, even on beefy hardware (hexacore) with plenty of RAM (16G), and
nothing else going on (single instance). I
On Thu, Sep 01, 2011 at 10:32:52AM -0400, Jeff LaCoursiere wrote:
> I tried and failed with VirtualBox too. Timing seemed impossible to
> maintain, even on beefy hardware (hexacore) with plenty of RAM (16G), and
> nothing else going on (single instance). I don't think VirtualBox is up
> to
> In our office, we were running an Asterisk 1.6.2.14 machine with DAHDI
> 2.3.0.1, FreePBX 2.8.1 and an analog DAHDI card with 8 FXO ports. This
> machine had several DAHDI trunks defined in the FreePBX interface,
> each one containing a single DAHDI channel. It
> also had a few outgoing routes de
In our office, we were running an Asterisk 1.6.2.14 machine with DAHDI 2.3.0.1, FreePBX 2.8.1 and an analog DAHDI card with 8 FXO ports. This machine had several DAHDI trunks defined in the FreePBX interface, each one containing a single DAHDI channel. It
also had a few outgoing routes defined in
Alessio wrote:
I have 2 computers in the lan, one is the Asterisk PBX and the other
is the server with hylafax and iaxmodem installed.
.
Sep 1 16:50:11 FAXServer FaxGetty[6225]: --> [4:RING]
Sep 1 16:50:11 FAXServer FaxGetty[6225]: ANSWER: Call ID 1 "06654321"
Sep 1 16:50:11 FAXServer Fax
On 09/01/2011 11:21 AM, Thorolf Godawa wrote:
Hi,
I've been testing the T.38 functionality in 10.0.0-beta1
with very successful results.
what about 1.8?
Will the T38 enhancement also be included in the 1.8 version?
T.38 gateway will not be included in the 1.8 releases. It was a feature
dev
Hi,
> I've been testing the T.38 functionality in 10.0.0-beta1
> with very successful results.
what about 1.8?
Will the T38 enhancement also be included in the 1.8 version?
Thanks a lot,
--
Chau y hasta luego,
Thorolf
--
_
-
Steve Underwood wrote:
On 09/01/2011 11:50 PM, Lee Howard wrote:
kirsten du toit wrote:
You should try disabling ecm..
This seems crazy to me. Why are you recommending it?
Because its the industry standard last resort of anyone who doesn't
understand FAX and is using T.38.
Even HP recomm
Hi!
from 2 days I'm trying to run hylafax server and iaxmodem with Asterisk 1.8.5.
I have 2 computers in the lan, one is the Asterisk PBX and the other is the
server with hylafax and iaxmodem installed.
In Asterisk I set up an IAX trunk in this way:
___
iax.conf
[iaxmode
On 09/01/2011 11:50 PM, Lee Howard wrote:
kirsten du toit wrote:
You should try disabling ecm..
This seems crazy to me. Why are you recommending it?
Because its the industry standard last resort of anyone who doesn't
understand FAX and is using T.38.
Steve
--
_
On Thu, Sep 1, 2011 at 9:25 PM, RSCL Mumbai wrote:
> On Thu, Sep 1, 2011 at 8:02 PM, Jeff LaCoursiere wrote:
>
>>
>> On Thu, 1 Sep 2011, RSCL Mumbai wrote:
>>
>> Hi,
>>>
>>> Anyone using Asterisk on Virtualbox.
>>>
>>> I am using and facing CPU peaking issue.
>>>
>>> Hardware is IBM X3200 M3, Q
On Thu, Sep 1, 2011 at 8:02 PM, Jeff LaCoursiere wrote:
>
> On Thu, 1 Sep 2011, RSCL Mumbai wrote:
>
> Hi,
>>
>> Anyone using Asterisk on Virtualbox.
>>
>> I am using and facing CPU peaking issue.
>>
>> Hardware is IBM X3200 M3, Quad Core Xeon 3 GHz with 4 GB RAM (2 cores
>> and 2 GM RAM allocat
kirsten du toit wrote:
You should try disabling ecm..
This seems crazy to me. Why are you recommending it?
Lee.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live i
On Thu, 1 Sep 2011, RSCL Mumbai wrote:
Hi,
Anyone using Asterisk on Virtualbox.
I am using and facing CPU peaking issue.
Hardware is IBM X3200 M3, Quad Core Xeon 3 GHz with 4 GB RAM (2 cores
and 2 GM RAM allocated to the asterisk VM -- thats the only VM as of
now), 64bit CentOS 5.4.
Only SIP
If I recall correctly, wav has headers and alaw does not. If this is so,
that would be where the problem lies.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
Sent: Thursday, September 01, 2011 9:
On Thu, 1 Sep 2011, RSCL Mumbai wrote:
Error Processing: "sox failed to convert file and original could not be
copied as a fall back" for wav_Track111.alaw!
Maybe FreePBX does not know how to construct a sox command line for files
that do not have headers.
--
Thanks in advance,
Hi,
Anyone using Asterisk on Virtualbox.
I am using and facing CPU peaking issue.
Hardware is IBM X3200 M3, Quad Core Xeon 3 GHz with 4 GB RAM (2 cores
and 2 GM RAM allocated to the asterisk VM -- thats the only VM as of
now), 64bit CentOS 5.4.
Only SIP and softphones.
Max 10 simultaneous calls.
On Thu, 1 Sep 2011, RSCL Mumbai wrote:
I simply convert all my audios (MOH, accouncements) to .wav format,
16bit, 11kHz (I believe this is the best format for asterisk).
8KHz?
(a) How can I check the codec format of my announcements, MOH ?
The 'file' command will show you the format for so
Surprisingly, despite the error message, the files is uploaded in
"/var/lib/asterisk/mohmp3" with correct permissions and ownership.
Its not showing in FreePBX MOH Screen.
I guess its a FreePBX issue.
Sans
On Thu, Sep 1, 2011 at 7:56 PM, RSCL Mumbai wrote:
> Thanks again @Danny.
>
> File conv
- Original Message -
> On 11-09-01 07:04 AM, Tim King wrote:
> > I have found numerous claims that 1.8 can do T.38 gateway with a
> > patch,
> > however I am yet to find the patch or any instructions on
> > implementing it.
> > Anyone have a link?
> >
> Asterisk-10.0.0-beta1 is another opti
Thanks again @Danny.
File converter worked like a charm.
asterisk -rx "file convert /var/lib/asterisk/mohmp3/wav_Track11.wav
wav_Track11.alaw"
I coped the new file from sounds/ folder to my desktop
And I tried to upload the new .alaw file using FreePBX,
I got the following error:
Error Processi
On 11-09-01 07:04 AM, Tim King wrote:
I have found numerous claims that 1.8 can do T.38 gateway with a patch,
however I am yet to find the patch or any instructions on implementing it.
Anyone have a link?
Asterisk-10.0.0-beta1 is another option.
--
Paul Belanger
Digium, Inc. | Software Develop
Asterisk has a built-in file convert
help file convert
Usage: file convert
Convert from file_in to file_out. If an absolute path is not given, the
default Asterisk sounds directory will be used.
Example:
file convert tt-weasels.gsm tt-weasels.ulaw
-Original Message-
From: aster
Thx @Danny
I am feeling a bit lost here...
We are using G711-aLaw for all our calls (endpoints) and I would like
to align everything to this codec.
I have an MOH file -- a custom wav file. How do I check its codec format ?
And if its not G711-aLaw, how do I convert it to G711-aLaw.
Thank you.
Maybe this will be better than my first answer – Audio files do indeed have
codec formats. If you are in a particular codec (say G729),
Playback/Background and MOH will search for files that match the codec format
first, then transcode WAV/GSM/whatever you have to that format if it isn’t
foun
Hi Tim,
On 09/01/2011 03:49 AM, Tim King wrote:
I realize that faxing is not great with voip but here is my confusion.
I have been working on a web based fax system for 2 weeks. During this
time I have sent over 100 2 page faxes without any errors. Now today
as things are finally completed I c
Hi,
In my Office all our users have a Desk phone. Some of the users who are
using laptops have a Soft phone too along with their Desk phone. Right now
we are using two different extensions for their desk and soft phones.
Is it possible to have simultaneous ring for both the extensions (ie. soft
p
On 1/09/2011 7:04 PM, Tim King wrote:
I have found numerous claims that 1.8 can do T.38 gateway with a
patch, however I am yet to find the patch or any instructions on
implementing it. Anyone have a link?
https://issues.asterisk.org/view.php?id=13405
--
___
I have found numerous claims that 1.8 can do T.38 gateway with a patch,
however I am yet to find the patch or any instructions on implementing it.
Anyone have a link?
--
_
-- Bandwidth and Colocation Provided by http://www.api-digi
That debug looks cool but I have no idea what it means.
If you are using T.38, turn it off, and do audio fax, recorded with MixMonitor.
When you can hear the audio of the fax hopefully you will be able to
tell what's going on, and if you're lucky it's something specific to
the particular kind of
You should try disabling ecm..
put the following in res_fax.conf
; Enable/disable T.30 ECM (error correction mode) by default.
; Default: Enabled
ecm=no
On Thu, Sep 1, 2011 at 1:27 AM, C F wrote:
> I think you should change the subject line to:
> Faxes suddenly worked for 2 weeks.
>
> On Wed
Hi,
Does audio files have codec formats? I simply convert all my audios (MOH,
accouncements) to .wav format, 16bit, 11kHz (I believe this is the best
format for asterisk).
I am new to this and may be incorrect.
Going forward,
(a) How can I check the codec format of my announcements, MOH ?
(b) How
Bruce, that's exactly the command I was looking for.
Thx a ton.
Sans
On Thu, Sep 1, 2011 at 12:17 AM, Bruce B wrote:
> "sip show channels" is the command you are looking for.
>
>
> On Wed, Aug 31, 2011 at 2:45 PM, RSCL Mumbai wrote:
>
>> asterisk -rx "core show channels verbose" does not provi
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