Dears;
I am facing the following problem:
Already we requested from the service provider to enable the auto jumping
service for our analoge telephone lines, so because we have 4 telephone lines
from the service provider, then if you called line # 1 and it was busy, then
the call will be sent
On Wednesday 17 October 2012, bilal ghayyad wrote:
Dears;
I am facing the following problem:
Already we requested from the service provider to enable the auto jumping
service for our analoge telephone lines, so because we have 4 telephone
lines from the service provider, then if you
Hi,
Our client has DAHDI groups with 4 PRIs in each group (one 4-port
interface per group), up to 6 groups per server. When we dial, we can
specify the group to be used for dialling, and our dial plan
automatically distributes calls over multiple servers and multiple
groups within a server.
Dave Platt provided the following answer to a similar question of mine
last week. I was trying to use SoftHangup() to prempt a DAHDI line for
an emergency call. Here is his reply.
That may be due to a common characteristic of PSTN lines (at least,
it's common here in the U.S.)
By design,
In article 201210171813.45334.r...@linux-delhi.org,
Raj Mathur (राठमाथॠर) r...@linux-delhi.org wrote:
Hi,
Our client has DAHDI groups with 4 PRIs in each group (one 4-port
interface per group), up to 6 groups per server. When we dial, we can
specify the group to be used
On Wed, Oct 17, 2012 at 8:43 AM, Raj Mathur (राज माथुर)
r...@linux-delhi.org wrote:
Hi,
Our client has DAHDI groups with 4 PRIs in each group (one 4-port
interface per group), up to 6 groups per server. When we dial, we can
specify the group to be used for dialling, and our dial plan
Hello,
I posted this problems in the past and was not able to find the solution, at
the time I posted this issue I had a equipment malfunction which has been
fixed but I still have the -- Requested transfer capability: 0x00 - SPEECH
error.
Any suggestions?
Asterisk 1.8.17.0 built by root @
motty.cruz wrote:
Hello,
Hola,
I posted this problems in the past and was not able to find the solution, at
the time I posted this issue I had a equipment malfunction which has been
fixed but I still have the -- Requested transfer capability: 0x00 - SPEECH
error.
That's not an error. Are
On Wednesday 17 Oct 2012, Tony Mountifield wrote:
In article 201210171813.45334.r...@linux-delhi.org,
Raj Mathur (à€°à€Ÿà€ à€®à€Ÿà€¥à¥ à€°) r...@linux-delhi.org wrote:
Our client has DAHDI groups with 4 PRIs in each group (one 4-port
interface per group), up to 6 groups per server. When
Thanks Joshua, Actually you're right, I'm not experiencing any issue. I
don't see that Requested transfer capability: 0x00 - SPEECH in older
version of Asterisk so I was wondering? And since my is from PRI to BPX.
Thanks
-Original Message-
From: asterisk-users-boun...@lists.digium.com
motty.cruz wrote:
Thanks Joshua, Actually you're right, I'm not experiencing any issue. I
don't see that Requested transfer capability: 0x00 - SPEECH in older
version of Asterisk so I was wondering? And since my is from PRI to BPX.
The message is only printed out if your verbose level is at a
I again would recommend a more thorough explanation of the configs
I've been using Asterisk for years - but the configs for this need some
explanation in the wiki
The samples contradict what the wiki has.. And as I indicated I could
not get audio working...
On 10/15/12 10:11 AM, Joshua Colp
Robert wrote:
I again would recommend a more thorough explanation of the configsŠ
The sample configuration details the various options of the channel
driver and some very simple generic examples, they aren't made to just
work for various services and clients.
I've been using Asterisk for
Actually I am not talking on how to handle it in the extensions.conf because I
am doing same as you wrote. But even so, I am facing a problem that some calls
are captured and some calls are not captured.
Currently, I set the callwaiting=no in the chan_dahdi.conf, it seems it is
working fine.
My company has been running Asterisk 1.6.2.19-1_centos5 from the official
yum repo, and for a while now I've been receiving complaints from our call
centers about calls not being routed in the most efficient order.
I'll explain with a simplified scenario--
Let's say I have two queues: A and B. I
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alex Forster
Sent: Wednesday, October 17, 2012 2:42 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Agents in more than one queue at once
My company has been
The Asterisk Development Team has announced the second release candidate of
Asterisk 11.0.0. This release candidate is available for immediate
download at http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 11.0.0-rc2 resolves several issues reported by the
community and
Hi,
With regards to:
On Mon, 2012-10-15 at 09:09 -0500, Joshua Colp wrote:
asterisk asterisk wrote:
Dear all,
Hola,
I wish to ask a question of the new Motif Channel in asterisk 11.
I successfully compile the binary and run without error. However, when
dialing out, no external
Hans Witvliet wrote:
And to: Asterisk 11.0.0-rc2 Now Available
skimming through
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.0.0-rc2
I did not see any reference towards Motif/XMPP.
So your code is still only in SVN, not in the RC2?
The commits in question:
* [r374850]
Does anyone on the list have any experience with using a Sangoma D500 card with
Asterisk to transcode G729? If you could mention pros and cons I would like to
hear opinions.
Thanks
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
--
Dear All,
I've got this Warning message on my log:
WARNING[3741]: res_phoneprov.c:923 set_config: Unable to load users.conf
what is this mean? thank you.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com
On Thu, 18 Oct 2012, Rizha Yuherdianto wrote:
I've got this Warning message on my log:
WARNING[3741]: res_phoneprov.c:923 set_config: Unable to load users.conf
what is this mean? thank you.
I'm just a 1.2 Luddite, but I'd guess:
0) It's just a warning so it may not be a big deal.
1) The
Thank you Steve,
I'm using AsteriskNow latest version 2.0.2.
my answer is:
0) if its just a warning, how to get it fixed?
1) checked, it is not exist. is it exist by default?
2) what directory it should be?
3) im root
4) the file doesn't exist
Thanks
On Thu, Oct 18, 2012 at 11:18 AM,
On Thu, 18 Oct 2012, Rizha Yuherdianto wrote:
0) if its just a warning, how to get it fixed?
It doesn't really need to. A 'warning' is like saying here's something
you should be aware of.
Personally, I prefer to resolve all warnings so there is less cruft to
sift through when something
0) if its just a warning, how to get it fixed?
It doesn't really need to. A 'warning' is like saying here's something
you should be aware of.
Personally, I prefer to resolve all warnings so there is less cruft to
sift through when something actually does go wrong.
I see
1)
On Thu, Oct 18, 2012 at 1:35 AM, Jared Baxley jared.bax...@gmail.comwrote:
I was following Digium's instructions to the letter to install g729. but
upon telling asterisk to load the module, the system hung.
after a few minutes later a CTRL-C and attempted to run the command again.
Same
On Thu, Oct 18, 2012 at 1:49 AM, Steve Totaro
stot...@totarotechnologies.com wrote:
On Thu, Oct 18, 2012 at 1:35 AM, Jared Baxley jared.bax...@gmail.comwrote:
I was following Digium's instructions to the letter to install g729. but
upon telling asterisk to load the module, the system hung.
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