the link for AMI 1.3
specification ?
Unfortunately, there was no AMI specification previous to AMI 1.4.
--
Rusty Newton
Digium, Inc. | Community Support Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
direct: +1 256 428 6200
Check us out at: http://digium.com http://asterisk.org
in the below CHANGES document:
http://svnview.digium.com/svn/asterisk/branches/12/CHANGES?revision=400742view=markup
--
Rusty Newton
Digium, Inc. | Community Support Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
direct: +1 256 428 6200
Check us out at: http://digium.com http
have asked about
it. :)
So.. can't write to function is definitely expected behavior.
Hope that helps!
--
Rusty Newton
Digium, Inc. | Community Support Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
direct: +1 256 428 6200
Check us out at: http://digium.com http://asterisk.org
more than 5 inserts to a database within a
few hours)
You can tell I haven't used func_odbc in ages. :) Glad you figured it
out despite me misleading you unwittingly. An Asterisk restart! It is
usually the simple things.
--
Rusty Newton
Digium, Inc. | Community Support Manager
445 Jan Davis
that Asterisk 12 is still in beta and not widely used by the
user-base yet, you may want to ask this on the asterisk-dev list.
Those working on Asterisk 12 features may have more insight,
especially since the scope of your question covers Asterisk internals
(2.A)
--
Rusty Newton
Digium, Inc. | Community
for something
more current.
I suspect RH5 and RH6 are most popular...but I'm looking for facts
I don't have any numbers, but I watch the issue tracker a lot and I
see pretty much CentOS, Debian and Ubuntu. Which seems to match what
everyone else is saying on this thread.
--
Rusty Newton
Digium, Inc
at email attached to this messages.
Regards,
Eddie
In the future please only use the asterisk-biz list for job postings.
See http://www.asterisk.org/community/discuss for more info on the
lists,IRC channels and their topics.
Thanks,
--
Rusty Newton
Digium, Inc. | Community Support Manager445
On Fri, Sep 27, 2013 at 3:20 AM, Gareth Blades
mailinglist+aster...@dns99.co.uk wrote:
On 26/09/13 16:43, Rusty Newton wrote:
On Thu, Sep 26, 2013 at 10:08 AM, Gareth Blades
mailinglist+aster...@dns99.co.uk wrote:
On 26/09/13 14:59, Rusty Newton wrote:
Try the following:
extension
On Fri, Sep 27, 2013 at 10:20 AM, Paul Albrecht palbre...@glccom.com wrote:
What's up with the user list archive? It hasn't been updated since the 23rd.
We did some Mailman maintenance on the 23rd, some configuration may
have been goofed up. We'll look into it. Thanks!
--
Rusty Newton
Digium
that you must use the bindaddr=::
for it to listen for IPv4 IPv6 but when I do this my IPv4 connections
drops.
You responded to the wrong thread. oops! :)
--
Rusty Newton
Digium, Inc. | Community Support Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: http
))
The argument passed to the r option should be the specific tone in the
category of the tonezone you are setting.
--
Rusty Newton
Digium, Inc. | Community Support Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: http://digium.com http://asterisk.org
On Thu, Sep 26, 2013 at 10:08 AM, Gareth Blades
mailinglist+aster...@dns99.co.uk wrote:
On 26/09/13 14:59, Rusty Newton wrote:
Try the following:
extension = 6001,1,Set(CHANNEL(tonezone)=us)
same = n,Dial(SIP/6001,,r(ring))
The argument passed to the r option should be the specific tone
!
These messages alone don't show the whole picture.
https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information
Collect a log with VERBOSE and DEBUG turned up to level 5, SIP debug
turned on, and pastebin that.
I'd wait until after you test with the latest in 1.8
--
Rusty Newton
'^;' sample.conf sample.conf.nocomments)
4. Also, pastebin a log with DEBUG turned up
https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information
--
Rusty Newton
Digium, Inc. | Community Support Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
direct: +1 256 428 6200
Check
open on the firewalled system.
--
Rusty Newton
Digium, Inc. | Community Support Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
direct: +1 256 428 6200
Check us out at: http://digium.com http://asterisk.org
into.
--
Rusty Newton
Digium, Inc. | Community Support Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
direct: +1 256 428 6200
Check us out at: http://digium.com http://asterisk.org
--
_
-- Bandwidth and Colocation
? Were you eventually able to get the cert2 versions?
--
Rusty Newton
Digium, Inc. | Community Support Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: http://digium.com http://asterisk.org
. Sounds like 19 in this case.
So using Hangup(19) in your dialplan may send out the 480 at least.
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Rusty Newton
Digium, Inc. | Community Support Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: http://digium.com http://asterisk.org
lookups
since you are only pulling info and comparing it to what the user entered
through DTMF.
http://bloke.org/whmcs/whmcs-and-asteriskfreepbx-integration/
--
Rusty Newton
Digium, Inc. | Community Support Manager445 Jan Davis Drive NW -
Huntsville, AL 35806 - US
Check us out at: http
the output of core show version as well.
--
Rusty Newton
Digium, Inc. | Community Support Manager445 Jan Davis Drive NW -
Huntsville, AL 35806 - USdirect: +1 256 428 6200
Check us out at: http://digium.com http://asterisk.org
, then turn up DEBUG messages to
level 5 and see where the call is going at the moment of transfer.
--
Rusty Newton
Digium, Inc. | Community Support Manager445 Jan Davis Drive NW -
Huntsville, AL 35806 - US
Check us out at: http://digium.com http://asterisk.org
(asterisk-current)
Requires: dahdi-firmware
Can this please be fixed.
That's no fun! Thanks for reporting that. I'll ping the appropriate people to
see what's up.
--
Rusty Newton
OS Community Support Manager | Digium, Inc. | www.digium.com
- Original Message -
From: Rusty Newton rnew...@digium.com
- Original Message -
From: Patrick Lists asterisk-l...@puzzled.xs4all.nl
Hi,
The dahdi-firmware package seems to be missing in the
asterisk-current
repo on http://packages.asterisk.org
-- Finished
be able to answer this here, but the list is not intended for this
kind of discussion. In the future, please refrain from posting general VOIP or
equipment specific questions when unrelated to Asterisk.
--
Rusty Newton
OS Community Support Manager | Digium, Inc. | www.digium.com
of the reload with
WARNING,ERROR,NOTICE, plus VERBOSE and DEBUG turned up (try 5 or higher) then
maybe there will be something interesting..
--
Rusty Newton
OS Community Support Manager | Digium, Inc. | www.digium.com
information be provided to
the calling channel.
Yes. It may not make much sense depending on what is on the far end.
Is there a risk involved when using the Progress()-command ?
Maybe, but not that I can think of at the moment. :)
--
Rusty Newton
OS Community Support Manager | Digium, Inc
analog configs? Most of my
searches have wound me up with GUI folks, and I'm just doing good
ol-fashioned hand editing on an Ubuntu system.
http://docs.digium.com/TDM410/analog410series_manual.pdf see pages 38-44
Hope those help!
--
Rusty Newton
OS Community Support Manager | Digium, Inc
with VERBOSE and DEBUG. Sanitize it as needed, and then link
to a pastebin with a log excerpt covering from the very beginning of the
attempted call to the end.
You may also want to include your iax.conf configuration, sanitized too of
course.
--
Rusty Newton
OS Community Support Manager | Digium
with VERBOSE and
DEBUG messages captured at the same time as a SIP packet capture including all
RTP as well.
Let us know what happens..
--
Rusty Newton
OS Community Support Manager | Digium, Inc. | www.digium.com
Office/Cell/Fax: 256-428-6200
that, you'll probably be able to figure it out.
But you probably want to know whether you already installed Asterisk via
packages.. as then the whole process may be different.
--
Rusty Newton
OS Community Support Manager | Digium, Inc. | www.digium.com
the DTMF digit.
Maybe someone will be able to take a look.
I'd also really recommend using ConfBridge which is newer than MeetMe. If you
switch to ConfBridge I'd recommend an upgrade to the latest 1.8.X as well.
--
Rusty Newton
OS Community Support Manager | Digium, Inc. | www.digium.com
Office
matching for 1234567890 and dial that?
Do you see DTMF debug for all those digits?
If you do end up trying ConfBridge - I've never used it in 1.8. Others have
made me aware that ConfBridge wasn't the best in 1.8, and that it's much better
in 10 or preferably 11.
--
Rusty Newton
OS Community
you would get the CallerID to update during a connected SIP
session. I'm no SIP expert, but Googling around - I don't think it's possible,
at least easily...
http://forums.asterisk.org/viewtopic.php?f=1t=71351p=136777
http://forums.digium.com/viewtopic.php?p=152753
--
Rusty Newton
OS Community
for your needs.
[1] http://ofps.oreilly.com/titles/9781449332426/asterisk-SysAdmin.html
[2]
http://ofps.oreilly.com/titles/9781449332426/asterisk-DP-Deeper.html#confbridgeConferencing
[3] http://ofps.oreilly.com/titles/9781449332426/asterisk-DeviceStates.html#SLA
--
Rusty Newton
OS Community
notices :S
I'm not involved in the build process for RPMs, but it sounds like they are
waiting on the dahdi-linux 2.6.2 release to finish the new set of RPMS. I'd
throw out an estimate of 1-3 weeks.
--
Rusty Newton
OS Community Support Manager | Digium, Inc. | www.digium.com
to the old way please?
Thanks Andrew for pointing this out to me.
It's operating as it always has. Mailman is set to insert the list address in
the Reply-To field without stripping the original.
--
Rusty Newton
OS Community Support Manager | Digium, Inc. | www.digium.com
On 11/12/2012 5:05 AM, Lenz Emilitri wrote:
Hello all,
anybody knows if the PDFs for presentations held at Astricon 2012 are
available somewhere? I looked at the website but cannot find anything.
Thanks
l.
I'll try to find out today or tomorrow and post the answer on the list.
--
Rusty
of commercial support options.
Thanks,
--
Rusty Newton
Digium, Inc | Open Source Community Support Manager
Check us out at: www.digium.com www.asterisk.org
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com
you have the serial number of your card when you call (printed on the
card itself, and included with the documentation you received upon
purchase).
http://www1.digium.com/en/support/contact
Please don't cross-post across the lists.
Thanks,
--
Rusty Newton
Digium, Inc | Open Source Community
questions about problems or configuration with your Digium card, you can
always call Digium support (included free with the card)..
http://www.digium.com/en/supportcenter/custom-communications-solutions/
--
Rusty Newton
Digium, Inc | Open Source Community Support Manager
Check us out
.
If you can get the packet trace for the same calls you gather log data
for, that would be best.
Thanks!
[1] https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information
--
Rusty Newton
Digium, Inc | Open Source Community Support Manager
Check us out at: www.digium.com
you seek are located there. Ctrl + f for ringtone
throughout the document.
Also, 16-bit, 16kHz mono .wav according to the wiki.
--
Rusty Newton
Digium, Inc | Open Source Community Support Manager
Check us out at: www.digium.com www.asterisk.org
.
Example
In this example:
The ringtone is identified as FancyRinger
Example Ringtone Configuration.
[fancyringer]
type=ringtone
alias=FancyRinger
filename=FancyRinger.wav
--
Rusty Newton
Digium, Inc | Open Source Community Support Manager
Check us out at: www.digium.com www.asterisk.org
it appears you have) and you need to make sure a callerid= Blah
1234 is set for those extensions.
Also see the Connected Line Updates section at
https://wiki.asterisk.org/wiki/display/DIGIUM/Digium+phones+when+used+with+the+DPMA
--
Rusty Newton
Digium, Inc | Open Source Community Support Manager
-dev or asterisk-dev mailing list.
--
Rusty Newton
Digium, Inc | Open Source Community Support Manager
Check us out at: www.digium.com www.asterisk.org
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New
and then 2 again.
Thanks,
--
Rusty Newton
Digium, Inc | Open Source Community Support Manager
Check us out at:www.digium.com www.asterisk.org
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New
, and follow
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines
before posting it to the tracker.
--
Rusty Newton
Digium, Inc | Open Source Community Support Manager
Check us out at: www.digium.com www.asterisk.org
Sales
Fb.com/muneebiqbal
Skype: JustMuneeb
This is type of post (advertising or solicitation of business) is not
allowed on the asterisk-users mailing list. Any further posts like this
will result in an unsubscribe and ban.
Thanks,
--
Rusty Newton
Digium, Inc | Open Source Community Support
On 7/19/2012 10:50 AM, Muneeb Iqbal wrote:
My apologies Rusty
Will be careful in future.
Regards,
Muneeb Iqbal - (Ricky)
Manager Sales
Fb.com/muneebiqbal
Skype: JustMuneeb
Thanks. I forgot to mention that asterisk-biz is the appropriate place
for that type of post.
--
Rusty Newton
Digium
This issue has been resolved. The offender has been unsubscribed and banned.
Thanks for your patience!
--
Rusty Newton
OS Community Support Manager | Digium, Inc. | www.digium.com
Office/Cell/Fax: 256-428-6200
- Original Message -
From: Rusty Newton rnew...@digium.com
We realize there is an issue with a ticket system subscribed to the list
and responding directly to member's posts. We are working on resolving
the issue, please bear with us!
Thanks,
--
Rusty Newton
Digium, Inc | Open Source Community Support Manager
Check us out at: www.digium.com
or overall structure
between AsteriskNOW 1.7 and 2.0.X . Of course default software versions
and such have been upgraded, but it should contain the same packages you
mentioned and be just as simple to use overall.
--
Rusty Newton
Digium, Inc | Open Source Community Support Manager
Check us out
Kamlesh,
Out of curiosity, what was your driving reason for wanting to do this?
Thanks,
Rusty Newton
Open Source Community Support Manager | Digium, Inc | www.digium.com
Desk/Cell/Fax : 256-428-6200
On 5/17/2012 7:38 AM, Kamlesh Kumar wrote:
Hi,
can we load the settings of /etc/dahdi
SpanDSP, may be
on that list.
Thanks,
Rusty Newton
Open Source Community Support Manager
Digium, Inc | www.digium.com | www.asterisk.org
On 5/16/2012 12:44 PM, Sebastian Gutierrez wrote:
Hi,
I´m with asterisk 1.6.2.20
DAHDI Version: 2.5.0.2 Echo Canceller: HWEC, MG2
SpanDSP: spandsp
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