On Sat, 12 Feb 2011, ast guy wrote:
Hi,
I have been out of touch with asterisk for quit some time and needed some
recommendations. I am looking for SIP hardphone that works well with
asterisk server.
Of-course, there's always good old Grandstream... Read the archives for
lots of for's and
Hello
I'm using Asterisk 1.4.20, and can't have Asterisk not load modules I
don't need:
cat modules.conf
noload = codec_speex.c
ip04*CLI reload
ip04*CLI show modules
codec_speex.so
Just to check, I added the actual filename (.so):
cat
On 02/12/2011 10:53 PM, Mark Willis wrote:
Is it possible to do SIP-Asterisk-TDM in a single step with FFA? Or
does FFA always use TIFF files?
I'm using Free FFA on 1.6.2.15 and I want to be able to use SPA 2102
ATA's at the fax machines and send faxes directly over a PRI.
Asterisk does not
On 11-02-13 09:52 AM, Gilles wrote:
I'm using Asterisk 1.4.20, and can't have Asterisk not load modules I
don't need:
Does someone know why Asterisk still loads modules even with the above
lines in modules.conf?
It looks like you're loading Asterisk, which loads all the modules, then
On 02/13/2011 09:36 AM, Leif Madsen wrote:
I'm not sure reload actually looks at modules.conf at that point. It
probably just reloads all the modules you have in memory, rather than
unloading everything, then parsing modules.conf and loading everything
in there back into memory (which I think
My take on this is to not skimp on the phones. This is how people
relate to the phone system you install. Good phones will, to them,
imply a good system. And vise-versa.
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On Sun, 13 Feb 2011 09:28:19 -0700, Joel Maslak wrote:
My take on this is to not skimp on the phones. This is how people
relate to the phone system you install. Good phones will, to them,
imply a good system. And vise-versa.
Life is just too sort to suffer through using a cheap phone.
The
Try Set instead of SetVar.
On Sat, Feb 12, 2011 at 9:59 PM, Dan Dan dani.mani...@gmail.com wrote:
Hi,
I am having trouble passing variables via the call files, here is my call
file via the php:
fputs($oSocket, Action: login\r\n);
fputs($oSocket, Events: off\r\n);
Dear
I had good experience with asterisk + spandsp for sending and receiving
fax, if your ip phone supports fax, you need asterisk only as g711(no vad)
gateway.
best
On Sun, Feb 13, 2011 at 7:00 PM, Kevin P. Fleming kpflem...@digium.comwrote:
On 02/12/2011 10:53 PM, Mark Willis wrote:
Is it
On Sun, 13 Feb 2011 10:36:43 -0500, Leif Madsen
leif.mad...@asteriskdocs.org wrote:
Try either restarting Asterisk to see if the modules still load (it
shouldn't).
Before doing the reload, I'd do a module unload chan_speex.so then do your
reload and see if that works.
Thanks for the tip, but I
On 2011-02-13 15:21, Gilles wrote:
noload = codec_speex.c
Try noload = codec_speex.so
Mark
--
Mark Willis
Star One Telecom
Office: 1-800-889-7001
Cell: 210 880 5097
http://staronetel.com
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-- Bandwidth and Colocation
On Sun, 13 Feb 2011 15:32:03 -0600, Mark Willis
marksli...@markwillis.net wrote:
Try noload = codec_speex.so
That dit it. However, I'm puzzled by the fact that the default
filenames in modules.conf all ended with .c instead of .so:
===
/etc/asterisk cat modules.conf
[modules]
On 2011-02-13 15:49, Gilles wrote:
On Sun, 13 Feb 2011 15:32:03 -0600, Mark Willis
marksli...@markwillis.net wrote:
Try noload = codec_speex.so
That dit it. However, I'm puzzled by the fact that the default
filenames in modules.conf all ended with .c instead of .so:
===
On 2/10/11 5:54 AM, Christian Gansberger christian.gansber...@accm.at
wrote:
Hello,
Maybe try that:
In your incoming isdn context:
[isdn-incoming]
exten = s,1,Set(TIMEOUT(digits)=3)
exten = s,2,WaitExten(2)
exten = s,3,Dial(SIP/operator...)
exten = 10,1,Dial(SIP/10)
exten =
On Sun, 13 Feb 2011 15:59:52 -0600, Mark Willis
marksli...@markwillis.net wrote:
The are .so in every modules.conf I've seen.
You're right. It looks like the modules.conf that came with the
Asterisk package is totally wrong:
www.voip-info.org/wiki/view/Asterisk+config+modules.conf
Thank you.
Thks, now I understand for your cooperation.TIA
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*-Developer DataBase
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*-www.jqmicrosistemas.com
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On Thursday 10 February 2011 12:33:40 Rodrigo Lang wrote:
2011/2/10 Tilghman Lesher tilgh...@meg.abyt.es
On Thursday 10 February 2011 06:13:38 Rodrigo Lang wrote:
I wonder if it is possible, without touching the source code, to
Asterisk save the cdr with date in unix time instead of the
Here's the messages log. There's a line that says ERROR: Unsupported DS E1
CHIP (00:00)
Feb 14 10:09:19 server14 kernel: [6011515.237242] dahdi: Telephony Interface
Registered on major 196
Feb 14 10:09:19 server14 kernel: [6011515.237246] dahdi: Version: 2.4.0
Feb 14 10:09:19 server14 kernel:
How would be the dialplan for this context from-lan ???
*---*
*-Edwin Quijada
*-Developer DataBase
*-JQ Microsistemas
*-Soporte PostgreSQL
*-www.jqmicrosistemas.com
*-809-849-8087
*---*
On Friday 11 February 2011 16:37:49 Danny Nicholas wrote:
Hi gang,
In 500 words or less (if possible), please explain what is a
legal music-on-hold file? My boss hates the stuff provided with the
distribution and I figure that I'm asking for trouble if I take my Les
Mis tracks
Hi everyone,
I know it's off topic from Asterisk directly but yet related.
What sources do you use to limit SIP connecting customers to specific
countries by IP (e.g. allowing USA and not China). It would help me a lot of
you can note the sources you trust that are complete and up to date.
On Mon, 14 Feb 2011, Bruce B wrote:
What sources do you use to limit SIP connecting customers to specific
countries by IP (e.g. allowing USA and not China). It would help me a
lot of you can note the sources you trust that are complete and up to
date.
I compiled this list a few (6?) months
One possible advantage of the fact that IANA has depleted its pool of
/8's (class A) is that if you are only filtering at that level the data
is static now. It should never change again for IPV4.
John
On 2/13/2011 11:54 PM, Steve Edwards wrote:
On Mon, 14 Feb 2011, Bruce B wrote:
What
On Sun, 2011-02-13 at 22:54 -0800, Steve Edwards wrote:
On Mon, 14 Feb 2011, Bruce B wrote:
What sources do you use to limit SIP connecting customers to specific
countries by IP (e.g. allowing USA and not China). It would help me a
lot of you can note the sources you trust that are
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