On 10/30/09 12:05, Joseph wrote:
I just jumped to asterisk-1.6.1.8 and I calls will not go through to my
asterisk.
Same setup with asterisk-1.4 and calls get accepted.
sip show registry (asterisk-1.6):
Host dnsmgr Username Refresh State
sip.actio.pl:5060 N 4589835
- Eckhard Jokisch e.joki...@orange-moon.de wrote:
Hi,
I set up my asterisk so it does dial out on DAHDI/1 which is on an FXS
modul.
But the phone that is attached to the line does nothing at all.
asterisk-CLI shows a lot of
-- DAHDI/1-1 is ringing
-- DAHDI/1-1 is ringing
--
On Mon, Oct 12, 2009 at 05:59:16PM +0200, Eckhard Jokisch wrote:
Hi,
I set up my asterisk so it does dial out on DAHDI/1 which is on an FXS modul.
But the phone that is attached to the line does nothing at all.
asterisk-CLI shows a lot of
-- DAHDI/1-1 is ringing
-- DAHDI/1-1 is ringing
Sriram escribió:
Hi ,
I;ve Asterisk 1.6.0 with static agents (sip softphones with extns 100
101 ) in a queue..When a caller arrives in queue , it lands on
first 100 , 100 then does a blind transfer to 101 .. so that the
caller can converse with 101 .. strangely enough the queue_log
Kevin P. Fleming
Envoyé : vendredi 21 août 2009 17:11
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : Re: [asterisk-users] Asterisk 1.6 t 38 passthrough
BERGANZ François wrote:
I have that problem:
[Aug 21 15:57:35] WARNING[5198]: chan_sip.c:6737 get_ip_and_port_from_sdp
BERGANZ François wrote:
How can I do t-38 passthrough with asterisk 1.6 ?
I know how to do with 1.4 but not with 1.6…
There is no difference, the identical configuration should work. I would
recommend using the 1.6.0.14 or 1.6.1.5 release candidates (or any later
releases) as they contain a
-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] De la part de Kevin P. Fleming
Envoyé : vendredi 21 août 2009 15:05
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : Re: [asterisk-users] Asterisk 1.6 t 38 passthrough
BERGANZ François wrote:
How
BERGANZ François wrote:
When I receive a fax it is in g711
After pickup, the fax invite again with T38 in the SDP.
Have I something to insert in the dialplan or other to let the T38
passthrough ?
No.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive
-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] De la part de Kevin P. Fleming
Envoyé : vendredi 21 août 2009 15:31
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : Re: [asterisk-users] Asterisk 1.6 t 38 passthrough
BERGANZ François wrote:
When I receive
BERGANZ François wrote:
I have that problem:
[Aug 21 15:57:35] WARNING[5198]: chan_sip.c:6737 get_ip_and_port_from_sdp:
Failed to read an alternate host or port in SDP. Expect audio problems
[Aug 21 15:57:35] WARNING[5198]: chan_sip.c:17425 handle_request_invite:
Failed to set an
Hello D Tucny,
Your solution works indeed well, thanks for it:)
pepesz
Monday, August 3, 2009, 6:20:39 AM, you wrote:
2009/7/31 pepesz76 pepes...@o2.pl
Dear All,
I'n trying to make a simple call forwarding, however I have small
problem when evaluating an expresion.
Here is my
2009/7/31 pepesz76 pepes...@o2.pl
Dear All,
I'n trying to make a simple call forwarding, however I have small
problem when evaluating an expresion.
Here is my extensions.conf
...
; Unconditional Call Forward
exten = _#21*X.,1,Set(DB(CFIM/${CALLERID(num)})=${EXTEN:4})
exten =
On Tue, Jun 30, 2009 at 9:21 AM, Deric Pagederic.p...@nisc.coop wrote:
I've set up an outbound .call system for customer callbacks and the like.
Calls are going out over analog lines and I'm trying to use the
WaitForSilence routine to make sure the phone has stopped ringing before
starting
On Wed, May 13, 2009 at 3:30 AM, Jon Schøpzinsky j...@firstcom.dk wrote:
We are having some problems using t.38 together with a Cisco voice router at
one of our providers end.
We are using the new digium asterisk fax module to generate the fax, and
when we use together with our internal
: Asterisk Users Mailing List - Non-Commercial Discussion
Emne: Re: [asterisk-users] Asterisk 1.6 T.38 generation towards a Ciscovoice
router
On Wed, May 13, 2009 at 3:30 AM, Jon Schøpzinsky j...@firstcom.dk wrote:
We are having some problems using t.38 together with a Cisco voice router at
one of our
On Wed, May 13, 2009 at 9:21 AM, Jon Schøpzinsky j...@firstcom.dk wrote:
I used wireshark to debug the problem, and I can see that the cisco equipment
is correctly sending t.38 packets to asterisk, and the whole re-invite
process is successful.
The problem is, that Asterisk discards the t.38
Thank you .. appreciated.
Best Regards,
--
SplatNIX IT Services :: Innovation through collaboration
- Tilghman Lesher tilgh...@mail.jeffandtilghman.com wrote:
On Tuesday 28 April 2009 15:35:14 --[ UxBoD ]-- wrote:
HI,
I am trying to setup CDR with ODBC and MySQL but get the
On Tuesday 28 April 2009 15:35:14 --[ UxBoD ]-- wrote:
HI,
I am trying to setup CDR with ODBC and MySQL but get the following error :-
[Apr 28 21:30:01] ERROR[14567]: cdr_odbc.c:133 odbc_log: Unable to retrieve
database handle. CDR failed.
I can successfully connect with iSQL so ODBCINST
MaxGao wrote:
hi,all
i have just set up asterisk 1.6.0.7 rc1 with spandsp 0.0.5 pre4 to
ReceiveFAX, link to a E1 (DE410P) using dahdi
this can receive the fax from E1 successfully, but i see many error
message in the log like this:
[Mar 16 09:24:38] ERROR[23540] channel.c: ast_read() called
asterisk 1.4.23.2 and spandsp 0.0.4 get the same error nowbut less times
than other version ...
[Mar 16 10:12:50] DEBUG[23749]: chan_dahdi.c:7115 do_monitor: Monitor doohicky
got event Alarm on channel 1
[Mar 16 10:12:50] DEBUG[23752]: chan_dahdi.c:4731 __dahdi_exception: Exception
on 11,
On Sun, Mar 15, 2009 at 9:57 PM, MaxGao ss...@126.com wrote:
and many times when reciving tax , the E1 card will down , all the channel
get red alarm...
[Mar 16 09:49:19] DEBUG[20928] chan_dahdi.c: Monitor doohicky got event
Alarm on channel 2
[Mar 16 09:49:19] WARNING[20928] chan_dahdi.c:
Olivier wrote:
Hi,
My setup is:
IPPhone1 --- Asterisk1 with B410P Patton 4638 --- Asterisk2 ---
IPPhone2
I want to evaluate Asterisk1 in TE/PtmP mode.
So, Patton box is configured in NT/PtmP (with 3 BRI links between both
systems).
Anyway, asterisk -rx pri show spans keeps
Mike wrote:
I've read some sources claiming that Asterisk does not need DAHDI for
timing in 1.6.1. Is this true? Searching the web, all I can find is
pages celebrating the fact but no actual documentation on which version
it was introduced in and how one would go about configuring an
Wilton Helm wrote:
I still am not quite on the same page with you, though. There are a lot of
commands that aren't function calls that go into various config files. The
most basic and obvious one is
exten
There must be a hundred of these and I don't know where they are listed with
Thanks all very much for the help pointers.
I've found all of the documentation on asterisk (especially 1.2-1.4) to
be more than adequate, and the voip-info wiki to be almost complete for many
things I've had to do in the past.
I also back in 2004 was able to bring up several high end large
I meant digium.com.
Yay for messups!
It's been one of those weeks.
Really.
New to Aserisk 1.6 and find the 'installation tutorials' seem low to non
existent.
You go to the main Asterisk page (digium.org) and really just old install
instructions for 1.2 are in the examples.
Download
you can use any 1.4 how to but just use dahdi (both modules and tools)
David
2009/1/27 Steve Gladden aster...@michiganbroadband.com
I meant digium.com.
Yay for messups!
It's been one of those weeks.
Really.
New to Aserisk 1.6 and find the 'installation tutorials' seem low to non
Hi Steve -
New to Aserisk 1.6 and find the 'installation tutorials' seem low to non
existent.
Welcome to Open Source!
Seriously, look at the README files accompanying asterisk, dahdi, and
libpri. They will give you compilation/installation instructions.
You can also search this list with
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Steve Gladden wrote:
Is 1.6 so cutting edge that I should not expect to find complete
documentation (yet)like I seem to be expecting very easily?
Most of what is applicable to 1.4 is applicable to 1.6. I'm running 1.6
without any hiccups -- YMMV.
New to Aserisk 1.6 and find the 'installation tutorials' seem low to non
existent.
I first looked at * about four months ago and rapidly came to the same
conclusion. Even with the O-Reilly book, which I purchased in paper, although
it is freely downloadable, I feel there is a huge dearth of
On Tue, Jan 27, 2009 at 11:24:38AM -0700, Wilton Helm wrote:
New to Aserisk 1.6 and find the 'installation tutorials' seem low to non
existent.
I first looked at * about four months ago and rapidly came to the same
conclusion. Even with the O-Reilly book, which I purchased in paper,
Wilton Helm wrote:
[snip]
My conclusion after installing a worthless * demo (that actually does
allow two SIPs to talk to each other) is that Asterisk is not of any
value to anyone other than a person who makes a full time career out
of running Asterisk systems. I've installed and
Wilton Helm wrote:
[snip]
My conclusion after installing a worthless * demo (that actually does
allow two SIPs to talk to each other) is that Asterisk is not of any
value to anyone other than a person who makes a full time career out
of running Asterisk systems. I've installed and
Thanks for the reply. I have looked at the links you provided and I think they
will be useful. I may have some issues with drivers for the HFC, but I guess I
won't know until I try it.
Wilton
___
-- Bandwidth and Colocation Provided by
YMMV. Mine certainly did. For the better.
My comments were more negative than I intended. My installation is worthless
at this point because it is only a cookbook example and I haven't tried to
modify it to meet my needs. I didn't intend to imply that Asterisk is
worthless, just that I've
On Jan 27, 2009, at 10:50 PM, Wilton Helm wrote:
YMMV. Mine certainly did. For the better.
My comments were more negative than I intended. My installation is
worthless at this point because it is only a cookbook example and
I haven't tried to modify it to meet my needs. I didn't intend
I'm impressed that you picked up 6502 assembly out of an even larger
vaccum considering there was no 'net back then to help at all. Did
you install a PBX on an Atari?
No, I interfaced a Rockwell AIM to a 300 station Philips electromechanical PABX
(designed and built about 100 interface cards,
On Tue, Jan 27, 2009 at 12:50:42PM -0700, Wilton Helm wrote:
I just got a very nice posting from Tzafir showing me a web domain
I didn't even know existed.
It only includes documentation generated by 'make docs' . And is
actually linked from the README itself.
I'm not abandoning it by any
It actually does contain references of all applicaitons, CLI commands, and
such.
Where? I saw some examples, but I've never found an organized list of
commands. I'd love it.
Wilton
___
-- Bandwidth and Colocation Provided by
On Tuesday 27 January 2009 15:05:57 Wilton Helm wrote:
It actually does contain references of all applicaitons, CLI commands, and
such.
Where? I saw some examples, but I've never found an organized list of
commands. I'd love it.
For applications, Appendix B, and for dialplan functions,
Thanks for engaging with me on this. I picked up the book and I see what you
mean about Appendix B. I had under-appreciated it probably because of a
paradigm shift I need to make. I think you meant Appendix E rather than F for
dialplan.
I still am not quite on the same page with you,
Wilton Helm wrote:
Thanks for engaging with me on this. I picked up the book and I see
what you mean about Appendix B. I had under-appreciated it probably
because of a paradigm shift I need to make. I think you meant
Appendix E rather than F for dialplan.
I still am not quite on the
2009/1/17 Steve Gladden aster...@michiganbroadband.com
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.
As you're using an IP connection, chances are you'll get issues with faxing
if you
On Sat, Jan 17, 2009 at 11:51 AM, Steve Gladden
aster...@michiganbroadband.com wrote:
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
What you need is a so called T38Gateway application.
there is a patch o the tracker which you might want to try:
http://bugs.digium.com/view.php?id=13405
klaus
Steve Gladden schrieb:
The scenario we have is fax send/recieve software that ONLY talks T38
and an asterisk box.
We have ITSP
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?
___
-- Bandwidth and Colocation Provided by
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
___
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
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 instances of the
script running at a time because it takes so long to run it
On Mon, Nov 24, 2008 at 10:28 AM, Jeffrey Phelps [EMAIL PROTECTED] 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
-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
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
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.
___
-- Bandwidth and Colocation Provided by
I found the problem. In the file /etc/asterisk/cdr_mysql.conf the
default setting is to change clid to callerid. I remarked the line and
it is all working.
[aliases]
start=calldate
;callerid=clid
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David
Klaverstyn
Sent:
So it seems we've got a first successful experience with 1.6.
Are there any other ?
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
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
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
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
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 / DAHDI level, so using
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
- 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
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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