On Tue, Jun 24, 2014 at 5:25 AM, arun kumar wrote:
> Hello All,
>
> I have a Digium Wildcard TE410P Quad-Span T1 Card, when I do connect
> T1 lines it goes in RED. When I do connect the same line on a different
> Server (Same Model T1 Card) it works fine. How do I examine/diagnose my T1
> Car
On Fri, Jun 6, 2014 at 9:03 AM, Eduardo Leones wrote:
>
> Guys, I have a problem. I have a queue on asterisk 1.8 that members are
> added dynamically via the AMI QueueAdd. When you run the CLI a
> "reload app_queue.so" all members who were in the queue disappear. This is
> a bug or some parameter
On Tue, May 27, 2014 at 12:31 PM, Sevana Oy wrote:
> Hi,
>
> How do you figure out if one of gateways in your network leads to voice
> quality loss f.e. due to transcoding? The point is that all VoIP metrics in
> this case remain the same
>
> Thanks!
> Sevana
> http://www.sevana.fi
>
>
For tr
Here are links to the Asterisk Wiki for CDR and SIP tables. I didn't find
extensions listed, but it's pretty simple and I can provide the structure
for that if needed, but it would be without a definitive source beyond me
having used it for years. :-)
https://wiki.asterisk.org/wiki/display/AST/M
It's possible. Might want to look through everything here:
http://www.voip-info.org/wiki/view/SMS
On Fri, May 16, 2014 at 11:08 AM, Jayson Devor wrote:
> Hello Everyone,
>
> We have an order for SMS messaging. Can you gents and ladies be kind
> enough to
> disclose if SMS is possible using A
On Mon, May 12, 2014 at 1:25 PM, Nick Olsen wrote:
> Hello All, Looking for a little guidance on Real Time Pattern Matching.
>
> We are attempting to block outbound 411 via when someone dials
> NXX-555-, The must common being NXX-555-1212. However, We have some
> outbound providers that consi
On Thu, May 8, 2014 at 4:42 PM, Kevin Larsen <
kevin.lar...@pioneerballoon.com> wrote:
> > From: Josh Metzger
>
> If I recall correctly, the only reason we didn't like the built in paging
> feature is that it would put a paging soft button on every phone whe
On Thu, May 8, 2014 at 10:22 AM, Kevin Larsen <
kevin.lar...@pioneerballoon.com> wrote:
> > From: Josh Metzger
>
> > I'm currently working with Asterisk 11.8.1 trying to get Multicast
> > RTP working (it's not) with some Polycom phones, and I'm really
I'm currently working with Asterisk 11.8.1 trying to get Multicast RTP
working (it's not) with some Polycom phones, and I'm really trying to
determine if Asterisk or the phones are the issue. I THINK it's Asterisk...
In extensions.conf I have a simple: "Page(MulticastRTP/basic/x.x.x.x:)
line,
I'm not sure exactly what your use case is, but you could execute a Dial()
and use the "M" option to execute a Macro (or "U" to execute a gosub).
>From there, the call routes into the macro/subroutine, and you can process
away. After all of that is completed in the macro/subroutine, you can set
"M
This may be a more phone-specific question, but figured I'd ask to see if
someone has experience with this. I have a SIP phone (Polycom) configured
with two lines registered to two different Asterisk servers. I have
successfully configured SIP subscriptions to watch a different phone
registered t
Instead of using CDR for this, could you get the info you need using
channel event logging (Asterisk CEL)? I have never used it myself - just
something I've run across in the past that seems like it might work for
this case:
https://wiki.asterisk.org/wiki/display/AST/CEL+Design+Goals
On Thu,
With such a low amount of calls per month and with the extreme memory
limitations, it might be easier to write a script to pull out the data and
generate a static html page. Run it daily / weekly / whenever you need it.
On Thu, Apr 24, 2014 at 8:28 AM, binary dreamer wrote:
> already logrotate
m: asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] On Behalf Of Josh Metzger
> Sent: Wednesday, April 23, 2014 2:35 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Help with a bug
>
> As a second poss
I think it's all a matter of personal taste. I think the logic for "add to
DNC" is extremely trivial and would be more complicated with an AGI. You
have your prompt playback/read, if they hit "1", head to the queue, if they
hit "2", it's a single dialplan line to put the info into the database an
As a second possible solution, instead of "Record", could you use
MixMonitor, then run "StopMixMonitor" and THEN do your Playback? That
should definitely make sure the recording file is closed and the file
handle released.
-Josh
On Wed, Apr 23, 2014 at 2:29 PM, Josh Metzge
How many seconds later does the file show up? Can you just throw in a
Wait() (maybe 1 or 2 seconds) and then do the Playback, or would even a
second or two of delay be an issue (or does it still not work)?
-Josh
On Wed, Apr 23, 2014 at 2:23 PM, CDR wrote:
> Dear friends
> I filed a bug
> htt
I've always done my DB access via func_odbc and not with the mysql
package. While we ran a MySQL db, I was more comfortable with the odbc
stuff because it was part of Asterisk core and not an addon package. I
can't speak to the simplicity of using the mysql stuff vs the odbc stuff,
but there isn'
I agree that ODBC is the way to go here. It's trivially easy to setup, and
equally simple to push database updates via the dialplan. I've used ODBC
connectivity with Asterisk in a large and VERY busy call center, and
performance was never remotely an issue (call recording is a different
story, bu
To do it without using an external database, you could create a shell
script to do it that you would execute with a system call. You could get a
list of everything with:
asterisk -rx "database show"
>From there, you could grep the results for the value you are looking for,
use awk or even cut to
Try starting Asterisk with the -f option. It will NOT fork into the
background so you will see all messages on startup (including any that
might not end up in the log file). Search for DAHDI errors which will
likely be there.
Also, if you configure everything and start DAHDI but don't start Aste
I would modify the suggestions slightly and either add to it, or replace
the reference to voip-info with a link to https://wiki.asterisk.org
I was just thinking yesterday that voip-info seems out of date for many
things I search for, but at this point I'm usually looking up things that
are probabl
A little while back, I upgraded some systems from Asterisk 1.4 (I believe
it was 1.4.29.1, and they were already running DAHDI), to 11. 11 made the
most sense as far as it being a LTS release so it would be longer before
being forced into a new version (as far as continued bug fixes and things),
a
I've had years of experience using ODBC for CDR, SIP, and extensions with
Asterisk. One thing that has been problematic in the past is with
documentation as far as database tables changing between versions (even
within minor releases, though that was back in the 1.4 days). I was
excited to see th
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