The scenario we have is fax send/recieve software that ONLY talks T38
and an asterisk box.
We have ITSP providers that do NOT talk T38 but G711 only.
Does asterisk have the capability to take the T38 call from an ATA
or T38 software then bridge/transcode it and do G711 out to the PSTN
providers?
Try
module reload pbx_lua.so
Dominique Dartois wrote:
Hello all.
I'm playing with LUA and I can't see a way to reload 'extensions.lua' after
a change, except by restarting Asterisk.
Any clue?
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Great!!
Thanks a lot.
Try
module reload pbx_lua.so
Dominique Dartois wrote:
Hello all.
I'm playing with LUA and I can't see a way to reload 'extensions.lua'
after a change, except by restarting Asterisk.
Any clue?
---
Dominique Dartois
___
Hello all.
I'm playing with LUA and I can't see a way to reload 'extensions.lua' after
a change, except by restarting Asterisk.
Any clue?
Thanks.
-
Dominique Dartois
___
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On Tue, Nov 25, 2008 at 2:19 AM, Tilghman Lesher
[EMAIL PROTECTED] wrote:
On Monday 24 November 2008 11:38:09 am Atis Lezdins wrote:
On Sun, Nov 23, 2008 at 2:19 PM, Artifex Maximus [EMAIL PROTECTED]
wrote:
I've installed a new Asterisk 1.6.0.1 with addons and dahdi drivers
and tools but my
Hello!
On Tue, Nov 25, 2008 at 2:25 AM, Tilghman Lesher
[EMAIL PROTECTED] wrote:
On Monday 24 November 2008 06:19:44 pm Tilghman Lesher wrote:
On Monday 24 November 2008 11:38:09 am Atis Lezdins wrote:
On Sun, Nov 23, 2008 at 2:19 PM, Artifex Maximus [EMAIL PROTECTED]
wrote:
I've
: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Artifex
Maximus
Sent: martes, 25 de noviembre de 2008 11:25 a.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk 1.6 mysql cdr log problem
Hello!
On Tue, Nov 25, 2008 at 2:25 AM, Tilghman
] On Behalf Of Barry L. Kline
Sent: Sunday, 23 November, 2008 14:29
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Asterisk 1.6, IMAP Voicemail and externnotify
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
I have Asterisk sitting between the PSTN and a legacy PBX
or other purposes.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Barry L. Kline
Sent: Sunday, 23 November, 2008 14:29
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Asterisk 1.6, IMAP Voicemail and externnotify
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Jeffrey Phelps wrote:
I too am looking for a way to get the externnotify= script to run on
poll events.
Right now, I have a script that runs as a cron job every 60 seconds,
but with 150 voicemail boxes, I constantly have at least 40 or 50
On Sun, Nov 23, 2008 at 2:19 PM, Artifex Maximus [EMAIL PROTECTED] wrote:
Hi all!
I've installed a new Asterisk 1.6.0.1 with addons and dahdi drivers
and tools but my calls aren't logged. I'd enabled mysql log and
noticed that asterisk send a 'DESC cdr' so connection is working
between
On Monday 24 November 2008 11:38:09 am Atis Lezdins wrote:
On Sun, Nov 23, 2008 at 2:19 PM, Artifex Maximus [EMAIL PROTECTED]
wrote:
I've installed a new Asterisk 1.6.0.1 with addons and dahdi drivers
and tools but my calls aren't logged. I'd enabled mysql log and
noticed that asterisk
On Monday 24 November 2008 06:19:44 pm Tilghman Lesher wrote:
On Monday 24 November 2008 11:38:09 am Atis Lezdins wrote:
On Sun, Nov 23, 2008 at 2:19 PM, Artifex Maximus [EMAIL PROTECTED]
wrote:
I've installed a new Asterisk 1.6.0.1 with addons and dahdi drivers
and tools but my calls
Hi all!
I've installed a new Asterisk 1.6.0.1 with addons and dahdi drivers
and tools but my calls aren't logged. I'd enabled mysql log and
noticed that asterisk send a 'DESC cdr' so connection is working
between asterisk and mysql and I am able to call other phones so
Asterisk is working as
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
I have Asterisk sitting between the PSTN and a legacy PBX. Asterisk is
doing some IVR work prior to forwarding calls to the PBX and it also
acts as the voice mail server for the PBX, with Asterisk configured for
IMAP storage.
When a call comes in
On Wed, 2008-11-19 at 10:00 +1000, David Klaverstyn wrote:
Hi Guys,
Since moving to Asterisk 1.6, whenever I am using call files the call
is always logged with a disposition of NO ANSWER even though the call
is connected and answered. The duration displays the correct time.
Can
On Wed, 2008-11-19 at 13:34 -0700, Steve Murphy wrote:
On Wed, 2008-11-19 at 10:00 +1000, David Klaverstyn wrote:
Hi Guys,
Since moving to Asterisk 1.6, whenever I am using call files the call
is always logged with a disposition of NO ANSWER even though the call
is connected
Hi Guys,
Since moving to Asterisk 1.6, whenever I am using call files the call is
always logged with a disposition of NO ANSWER even though the call is
connected and answered. The duration displays the correct time. Can
anyone explain as to why when using call files the disposition is not
To follow up --
pbx_lua from trunk works as advertised when backported to 1.6.
pbx_lua from asterisk 1.6 seems hopelessly broken, and I've given up
on trying to persuade it to work.
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Howdy, all. I'm trying to use pbx_lua as included in Asterisk 1.6 --
but while it correctly reports an error on startup (but not reload!)
if extensions.lua does not exist, it doesn't appear to actually create
any contexts.
I'm testing in a very minimal configuration with autoload turned off;
: Monday, 27 October 2008 12:59 PM
To: Asterisk Users
Subject: [asterisk-users] Asterisk 1.6 CDR no Clid information
Hi All,
For some reason since moving to Asterisk 1.6. my CDR records are no
longer displaying the Clid field. The CDR records contain the Source
field be for some reason
Hi All,
For some reason since moving to Asterisk 1.6. my CDR records are no
longer displaying the Clid field. The CDR records contain the Source
field be for some reason not the CID details. I am logging CDR to
mysql.
Is anyone able to help?
Regards
David.
So it seems we've got a first successful experience with 1.6.
Are there any other ?
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AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
Hello users,
Can you share with me your experiences with Asterisk 1.6? Is it stable
enough for commercial service?
Thanks.
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AstriCon 2008 - September 22 - 25 Phoenix, Arizona
VoIP Cyprus wrote:
Can you share with me your experiences with Asterisk 1.6? Is it stable
enough for commercial service?
No. No matter how good some people may tell you it is, 1.6 is still
beta software and software is rarely beta for no good reason. Don't
even THINK about running 1.6
On 1 Sep 2008, at 17:34, Rob Hillis wrote:
VoIP Cyprus wrote:
Can you share with me your experiences with Asterisk 1.6? Is it
stable
enough for commercial service?
No. No matter how good some people may tell you it is, 1.6 is still
beta software and software is rarely beta for no good
Rob Hillis [EMAIL PROTECTED] writes:
No. No matter how good some people may tell you it is, 1.6 is still
beta software and software is rarely beta for no good reason.
Tell that to Google.
So far, for us, 1.6 beta is running better than any of the early 1.2
releases. Perhaps even better than
Hi List,
I recently switched to asterisk-1.6-beta9 because of the RPID support,
but ran into the Problem, that the RPID-Header is not sent.
sendrpid is set to yes in my sip.conf, and i'm even sure that the
add_header() function is called in chan_sip.c, but when i capture the
SIP-Packets,
Steve Totaro wrote:
I have consulted on so many systems with poor audio, the first thing I
check is IAX or SIP. If IAX, I move over to SIP and the calls are
prefect.
I avoid IAX at all costs, use OpenVPN, open tons of ports on your
firewall, whatever you can do to use SIP. The only time I
What model in the Polycom or Aastra range is the 360 level with?
2008/6/6 Chris Bagnall [EMAIL PROTECTED]:
When I pushed some vendors for prices there was only a tiny gap between
the 300 and 360. Would suggest looking hard at the 360 always...
Interesting... here in the UK the price
2008/6/7 Gavin Henry [EMAIL PROTECTED]:
What model in the Polycom or Aastra range is the 360 level with?
Probably the IP601:
http://www.voipon.co.uk/polycom-soundpoint-ip601-p-121.html
and 57i:
http://www.voipon.co.uk/aastra-57i-ip-phone-p-420.html
Snom 360:
On Thu, Jun 5, 2008 at 4:45 PM, Johansson Olle E [EMAIL PROTECTED] wrote:
5 jun 2008 kl. 20.45 skrev Michael Graves:
I wonder why more vendors haven't adopted IAX yet?
I expect that before major players adopt this protocol it'd need to be
confirmed as a standard by some form of
When I pushed some vendors for prices there was only a tiny gap between
the 300 and 360. Would suggest looking hard at the 360 always...
Interesting... here in the UK the price difference between the 300 and 360 is
pretty huge. Either you're getting some stunningly good pricing on 360s or
Tzafrir Cohen wrote:
On Thu, Jun 05, 2008 at 03:00:01AM +1000, Rob Hillis wrote:
Why on earth are you running two layers of echo cancellation - hardware
and software? To be honest, I think this is asking for trouble - I've
seen two occasions where having Oslec and hardware echo
On Thu, Jun 05, 2008 at 03:40:14AM +1000, Rob Hillis wrote:
Tzafrir Cohen wrote:
On Thu, Jun 05, 2008 at 03:00:01AM +1000, Rob Hillis wrote:
Why on earth are you running two layers of echo cancellation - hardware
and software? To be honest, I think this is asking for trouble - I've
Tzafrir Cohen wrote:
On Thu, Jun 05, 2008 at 03:40:14AM +1000, Rob Hillis wrote:
If you use a hardware EC (or technically: a span-specific echo
cancellation method) the generic Zaptel echo canceller (software-based,
OSLEC in this case) will not be used.
That's not always been my
On Thu, Jun 05, 2008 at 09:28:52PM +1000, Rob Hillis wrote:
Tzafrir Cohen wrote:
On Thu, Jun 05, 2008 at 03:40:14AM +1000, Rob Hillis wrote:
If you use a hardware EC (or technically: a span-specific echo
cancellation method) the generic Zaptel echo canceller (software-based,
Brent, hope your problems go away soon.
I am actually interested in the topic of this post. Ast 1.6 vs 1.4. We are
using asterisk 1.4.2 for a SIP only based configuration. Currently we have
about 200 SIP users which can cause approximately upto 3 simultaneous calls.
We are mainly concerned about
Hi!
I am actually interested in the topic of this post. Ast 1.6 vs 1.4. We
are using asterisk 1.4.2 for a SIP only based configuration. [...] We
are planning to accomodate about 5,000 users on this server.
Many people on this list will advise you to use a SIP proxy like
OpenSER in front of
Philipp von Klitzing wrote:
Hi!
I am actually interested in the topic of this post. Ast 1.6 vs 1.4. We
are using asterisk 1.4.2 for a SIP only based configuration. [...] We
are planning to accomodate about 5,000 users on this server.
Many people on this list will advise you to use a
I wonder why more vendors haven't adopted IAX yet?
I expect that before major players adopt this protocol it'd need to be
confirmed as a standard by some form of international body. That was
underway, but lacking anyone to push the process along.
I would've thought that Digium would be the most
On Thu, 2008-06-05 at 13:45 -0500, Michael Graves wrote:
I would've thought that Digium would be the most likely lead proponent,
but that doesn't seem to be the case.
Actually, Digium has been quite active in helping to try to get the IAX
protocol adopted as a standard. See
5 jun 2008 kl. 20.45 skrev Michael Graves:
I wonder why more vendors haven't adopted IAX yet?
I expect that before major players adopt this protocol it'd need to be
confirmed as a standard by some form of international body. That was
underway, but lacking anyone to push the process along.
Brent Davidson a écrit :
...I wonder why more vendors haven't adopted IAX yet?
Well, even ZoIPer (ex IdeFisk) team, still recommend using SIP over IAX
as SIP is more mature and reliable in asterisk and zoiper,
--
Benoit
begin:vcard
fn:Benoit Plessis
n:Plessis;Benoit
email;internet:[EMAIL
Is there some location that outlines the major differences between
Asterisk version 1.4 and version 1.6? I've read through change logs and
several technical discussions, but technical details don't really give
me the big picture. Basically, is 1.6 more stable than 1.4? Is it more
efficient?
2008/6/4 Brent Davidson [EMAIL PROTECTED]:
[snip]
We're currently using Asterisk 1.4.19, Zaptel 1.4.10,
Oslec SVN, Rhino R4FXO-EC cards, and Snom 300 Phones.
[snip]
Just a small aside...
You go to the trouble of building/using Oslec, and then use hardware
EC? Very odd. Does Oslec understand
Discussion
Subject: [asterisk-users] Asterisk 1.6 vs 1.4?
Is there some location that outlines the major differences between
Asterisk version 1.4 and version 1.6? I've read through change logs and
several technical discussions, but technical details don't really give
me the big picture. Basically
Brent Davidson wrote:
We're currently using Asterisk 1.4.19, Zaptel 1.4.10,
Oslec SVN, Rhino R4FXO-EC cards, and Snom 300 Phones.
Why on earth are you running two layers of echo cancellation - hardware
and software? To be honest, I think this is asking for trouble - I've
seen two occasions
On Thu, Jun 05, 2008 at 03:00:01AM +1000, Rob Hillis wrote:
Brent Davidson wrote:
We're currently using Asterisk 1.4.19, Zaptel 1.4.10,
Oslec SVN, Rhino R4FXO-EC cards, and Snom 300 Phones.
Why on earth are you running two layers of echo cancellation - hardware
and software? To be
Tzafrir Cohen wrote:
On Thu, Jun 05, 2008 at 03:00:01AM +1000, Rob Hillis wrote:
Brent Davidson wrote:
We're currently using Asterisk 1.4.19, Zaptel 1.4.10,
Oslec SVN, Rhino R4FXO-EC cards, and Snom 300 Phones.
Why on earth are you running two layers of echo cancellation -
Matt Watson wrote:
Have you tuned rxgain txgain in Zapata.conf? shameless-self-plug
http://www.mattgwatson.ca/2008/05/howto-tune-zaptel-dahdi-fxo-interfaces-on-asterisk-pbx/
/plug
Also, have you used fxotune to tune each FXO interface?
I believe echo cancellation happens at the Zaptel /
Just an update. I tried updating to the newest Rhino Release firmware
1.15 and newest stable driver version 2.2.6. It works OK with
zaptel-1.4.9.2 and compiles OK with 1.4.10.1 but when compiled against
zaptel 1.4.10.1 Asterisk does not see any zap channels. I'm currently
running one branch
On Fri, Mar 14, 2008 at 03:51:25PM +1100, Paul Hales wrote:
I just installed Asterisk 1.6 beta5 and moh is not working - is there a
trick? Or is something wrong with my system?
Could you please be more specific? An trace / config snippets of
whatever does happen?
--
Hi!
Paul Hales wrote:
I just installed Asterisk 1.6 beta5 and moh is not working - is there a
trick? Or is something wrong with my system?
This bug already fixed, you can check latest 1.6 branch or try to use
1.6 beta4. This version must not have this issue.
--
Best regards,
Igor
I just installed Asterisk 1.6 beta5 and moh is not working - is there a
trick? Or is something wrong with my system?
PaulH
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asterisk-users mailing list
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I am having issues with transfers (SIP/REFER) using Asterisk 1.6. You will
find the SIP debug below.
There are three phones in this setup. 5253 and 5258 are Aastra 53i telephones,
101 is a standard phone connected through an Audiocodes gateway. All phones
are registered in context phones
- Original Message
From: Jake Wicke [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com asterisk-users@lists.digium.com
Sent: Friday, 1 February, 2008 5:34:12 PM
Subject: [asterisk-users] Asterisk 1.6 - Problems with SIP/REFER
I am having issues with transfers (SIP/REFER
1 feb 2008 kl. 18.34 skrev Jake Wicke:
I am having issues with transfers (SIP/REFER) using Asterisk 1.6.
You will find the SIP debug below.
When you have issues, it's always a good idea to check the bug
tracker. There might be other people having the same issues, in some
cases,
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