To translate between g729 and g711 you need to buy some licences.
PaulH
- Original Message -
From: Lisa Wolf [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, February 16, 2006 11:38 AM
Subject: Re:
At 05:33 PM 02/15/2006, you wrote:
Most/all assemblers have a better and more consistent parser though.
The parser for the extensions dialplan is just short of insane
And the big advantage of most assemblers, manuals!
Ira
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Actually, you can do this:
exten = s,1,Set(TRUNK1=foo)
exten = s,n,Set(TRUNK2=bar)
exten = s,n,Set(TRUNK3=gak)
exten = s,n,Set(INDEX=1)
exten = s,n,Set(CURRTRUNK=${TRUNK${INDEX}})
exten = s,n,Dial(${CURRTRUNK}/555|60)
and you could increment INDEX (although these are local, (are you local?)
Hi all,
I've had a strange problem this morning and I know someone who has
reported exactly this problem to me too last week: -
I've setup a new server running Asterisk 1.2.4. Currently there is no
Zaptel hardware install (but there will be soon). This server is behind
a NAT router on an
Hello everyone.
This is a message I've sent before on Sunday, no one replied so I'm
reposting it (guess not everyone's at work 7/7)
I've got this really annoying and beyond-my-knowledge-to-debug problem. The
line connected to my FXO port gets marked out of order by my telco
operator. I don't
I am curious if anyone has had problems trunking iax2 with 100+
concurrent calls. I am planning on testing this out tomorrow, however I
wanted to know if anyone else has had a problem with this prior to my
test so I know what to look for if anything is known and what
resolutions have been found
Hi
Currently my
Asterisk is installed with default CDR settings. With these it is only
showing details for outgoing calls. All phone extens are in default
context.
Is it possible to
set up Call Detail Records for Inbound Calls? Would this be a change in
Manager_CDR.conf or other file?
A long time ago i tried to make one big iax2 trunk for one of my
customers, i soon changed this to several small trunks. (bandwith doesnt
rise all that much if you use 2 trunks instead of 1.) Asterisk didnt
seem to like my big trunk very much (i don't remember how big it was,
but probably
While I've never actually tried exactly what you're doing below
(constructing a variable name from strings and other variables), it looks
like the variable substitution you're attempting is not being done properly.
Try something like:
exten = s,3,GotoIf($[ ${NUM${mainLoop}_CMD} = Dial ]?5:7)
On Thu, 2006-02-16 at 13:38 +0200, Zoa wrote:
A long time ago i tried to make one big iax2 trunk for one of my
customers, i soon changed this to several small trunks. (bandwith doesnt
rise all that much if you use 2 trunks instead of 1.) Asterisk didnt
seem to like my big trunk very much (i
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
This can easily be accomplished with AMP using the Users and Devices mode.
http://voipspeak.net/index.php?/content/view/49/28/
How can this be done without AMP? Using personal queue's and agents? I need
information's to get better
Hi,
I have already determine the reason why my incoming
got release after one ring. The telco that I am
connected is waiting for an immediate answer
supervision from my side. Is there anyway immediate
answer supervision be included on the ISDN messages.
Thanks
--- leonimar cape [EMAIL
Hi,
I just want to inquire which of the available h323
modules for asterisk is more stable and better
quality. My boss asked me to setup asterisk with and I
am having a hard time choosing which one should I
used.
Any advice and suggestion will be greatly appreciated
Thanks in advance!
I think, but am not sure, that with a lot of calls inside the trunk,
some calls seemed to go suddenly go outside of the trunk in one or more
directions, bursts of error messages appeared on the cli etc.
i didnt investigate it a lot more, my problems went away with splitting
them up in
On 12:50, Thu 16 Feb 06, Tomislav Par?ina wrote:
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
This can easily be accomplished with AMP using the Users and Devices mode.
http://voipspeak.net/index.php?/content/view/49/28/
How can this be done without AMP? Using personal queue's
leonimar cape wrote:
Hi,
I just want to inquire which of the available h323
modules for asterisk is more stable and better
quality. My boss asked me to setup asterisk with and I
am having a hard time choosing which one should I
used.
Any advice and suggestion will be greatly appreciated
I
On Thu, 2006-02-16 at 14:04 +0200, Zoa wrote:
I think, but am not sure, that with a lot of calls inside the trunk,
some calls seemed to go suddenly go outside of the trunk in one or more
directions, bursts of error messages appeared on the cli etc.
i didnt investigate it a lot more, my
Hehe. This will be the same person looking for a GUI,
not finding one built in to asterisk and cant
understand why.
--- Anthony Rodgers [EMAIL PROTECTED] wrote:
You'll likely find Asterisk itself even more of a
challenge then.
On Feb 15, 2006, at 1:29 PM, roswel ajf wrote:
hi,
The trunks were made to be maximum 60 simultaneous channels iirc.
I doubt seriously you will be able to do 600 simultaneous on any system.
(with or without trunking). (at least out of the box).
Zoa
trixter aka Bret McDanel wrote:
On Thu, 2006-02-16 at 14:04 +0200, Zoa wrote:
I think, but
Thanks for the info yusuf... Im gonna check it out...
Cheers!
--- yusuf [EMAIL PROTECTED] wrote:
leonimar cape wrote:
Hi,
I just want to inquire which of the available h323
modules for asterisk is more stable and better
quality. My boss asked me to setup asterisk with
and I
am
Hi list,
any success trying to let internal calls ring differently than external
calls on a Grandstream BT102?
My settings, phoneside:
Default Ring Tone:system ring tone
x custom ring tone 1, used if incoming caller ID is *
custom ring tone 2, used if incoming caller ID is #
On Thu, 2006-02-16 at 14:29 +0200, Zoa wrote:
The trunks were made to be maximum 60 simultaneous channels iirc.
I doubt seriously you will be able to do 600 simultaneous on any system.
(with or without trunking). (at least out of the box).
Zoa
At 100 with g.729 its running 95% idle, in
When you have a lot of calls, try doing a show channels and iax2 trunk
debug. (they are killers)
Zoa
trixter aka Bret McDanel wrote:
On Thu, 2006-02-16 at 14:29 +0200, Zoa wrote:
The trunks were made to be maximum 60 simultaneous channels iirc.
I doubt seriously you will be able to do
hi
i am using asterisk-1.2.4 + asterisk-addons-1.2.1 on
2.6 kernal. i have added user in sip_buddies and
followed
http://www.voip-info.org/wiki-Asterisk+RealTime+Sip
but my ip phone is not registring properly.
asterisk is just sending SIP/2.0 404 Not found. i
think it must check DB table for
hi
i am using asterisk-1.2.4 + asterisk-addons-1.2.1 on
2.6 kernal. i have added user in sip_buddies and
followed
http://www.voip-info.org/wiki-Asterisk+RealTime+Sip
but my ip phone is not registring properly.
asterisk is just sending SIP/2.0 404 Not found. i
think it must check DB table for
trixter aka Bret McDanel wrote:
On Thu, 2006-02-16 at 14:29 +0200, Zoa wrote:
The trunks were made to be maximum 60 simultaneous channels iirc.
I doubt seriously you will be able to do 600 simultaneous on any system.
(with or without trunking). (at least out of the box).
Zoa
At 100 with
How can this be done without AMP? Using personal queue's and
agents? I need information's to get better picture about this one.
AMP doesn't do miracles! Look at its dialplan.
Mimmus
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On Thu, 2006-02-16 at 14:54 +0200, Zoa wrote:
When you have a lot of calls, try doing a show channels and iax2 trunk
debug. (they are killers)
Zoa
not having trunks set up yet, I dont do the latter but I do the former
all the time. Mostly becuase this is a new server and I wanted to make
Anybody out there using GR303? Latest grumblings on the list are from
last spring. I'd like to use Asterisk as a concentrator/DLC speaking
303 to a 5E. Threads from awhile back mentioned support only for
Asterisk spekaing to a concentrator, rather than acting as one itself,
but the
Kamran Ahmad wrote:
hi
i am using asterisk-1.2.4 + asterisk-addons-1.2.1 on
2.6 kernal. i have added user in sip_buddies and
followed
http://www.voip-info.org/wiki-Asterisk+RealTime+Sip
but my ip phone is not registring properly.
asterisk is just sending SIP/2.0 404 Not found. i
think it must
Progressinband=no fixed the issue for me. I've been onto Aastra support
already about it.
Lee
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Edward de
Zeeuw
Sent: 14 February 2006 14:55
To: Asterisk Users Mailing List - Non-Commercial Discussion
On Thu, 2006-02-16 at 14:58 +0200, yusuf wrote:
also doing IAX2 trunking. What do yuo mean you dont run asterisk out of
the box. Also want to know what is you bandwith usage for 100 calls and
g729
I run a modified version of asterisk. There are a few things that I
felt needed to be added,
Title: Firmware version 1.3.1 released for Aastra IP phones
There is no release note, just a text file that says
AASTRA TELECOM INC.
February 2006
FC-46-01-07.st - 9133i Generic SIP Firmware 1.3.1.1095
for customer release.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
Sometime my Budgetone 101 appears as UNKNOWN in 'sip show peers'.
I use 'qualify=1000' and my network is really stable. Why this?
Thanks
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Hello,
I am trying to write an AGI in Perl and I need to execute a function upon
answer of a call.
I know that there is the possibility to use the M() option in the Dial command
in order to do what I need, but that would mean that I would have to
incorporate the function's work in a
HI:
what is command on console to know how many calls are
sending in the same time?
__
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This is a message I've sent before on Sunday, no one replied so I'm
reposting it (guess not everyone's at work 7/7)
I've got this really annoying and beyond-my-knowledge-to-debug problem. The
line connected to my FXO port gets marked out of order by my telco
operator. I don't know how to
No, not AGI. I'll give the '' a shot. See, I'm a self-taught 'nix guy
and I always tend to skip the basics and go right for the good stuff.
I'm googling simple stuff all the time. For example, I just learned
about 'top' yesterday...
Bob McDowell
-Original Message-
From: [EMAIL
hi All,
Why do i get this error when I click on outbound routing?
Warning: Missing argument 5 for addroute() in
/var/www/html/admin/functions.php on line 1313
Warning: Missing argument 5 for addroute() in
/var/www/html/admin/functions.php on line 1313
has anyone encountered this error before?
On Thu, 2006-02-16 at 13:28 +, Lee Archer wrote:
There is no release note, just a text file that says
AASTRA TELECOM INC.
February 2006
FC-46-01-07.st - 9133i Generic SIP Firmware 1.3.1.1095 for
customer release.
http://www.aastra.com/support/show_manuals.asp?p=241
On Thu, 2006-02-16 at 05:46 -0800, jonny hashem wrote:
HI:
what is command on console to know how many calls are
sending in the same time?
I will guess 'show channels'
--
Trixter http://www.0xdecafbad.com Bret McDanel
UK +44 870 340 4605 Germany +49 801 777 555 3402
US +1 360 207 0479
Thanks
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dave
Cotton
Sent: 16 February 2006 14:02
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Firmware version 1.3.1 released for
AastraIPphones
On Thu,
If the suggestions that have already been posted don't work, then I'd
suggest running ethereal (or whatever your favorite packet capture utility
happens to be), and using the resulting trace to see what is happening on
the wire. You should be able to see a hangup if the issue is coming from
Title: Firmware version 1.3.1 released for Aastra IP phones
Any chance of getting a config option in that allows you
set headset/speaker in the audio menu?
Lee
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Gareth
OwenSent: 15 February 2006 02:00To:
Well... correct except that there is no [sipdevice].. it is all done
through IP registration on the other person's end.So.. all I have
is the dial statement. Is there a way to set a variable or something
right before the dial? (To my knowledge there isn't).
On 2/15/06, yusuf [EMAIL
[EMAIL PROTECTED] is believed to have said:
Is it me, is it the evil fate, is it the BT102?
I even updated the firmware to Software Version:
Program-- 1.0.6.7Bootloader-- 1.0.1.0HTML-- 1.0.0.49VOC-- 1.0.1.0
TIA,
Aldo
The attempt to obtain custom ringtones I described in my
That worked! thanks Kurt...
On a side note...
Somehow, ALL of the phone calls ended up coming to my SIP phone---
I need to figure out the appropriate 'inbound routing'
such that all calls coming from the PRI router (extensions 6350 through
6399)
get sent directly to the right
Doug,
This sounds reasonably plausible. He just might be fooling the busy detect
routine, kinda like how a female voice can trigger DTMF detection.
Bob,
Yeah, I will dig through the Cisco docs for a way to attenuate his mic...
Good point.
BTW: when editing zapata.conf, does a reload
Matt,
I you dont define a sip user/peer and just use a dial, asterisk will
automatically use the codec that it prefers, in my experince whenever i
dial SIP without defining a sip user/peer it always dials g711alaw/ulaw.
So in sip.conf in [general] (which would set codec choice for ALL sip
Leonimar,
I cant tell from the error, but i know oh323 is picky about exact
versions being used. So in my case i had Asterisk CVS 19/07/2005, i
used openh323-v1_13_5-src.tar.gz, pwlib-v1_6_6-src.tar.gz,
asterisk-oh323-0.7.2-pre1.tar.gz
the README says:
o PWlib (Portable Text and GUI
POWER FLUCTUATIONS I have in abundance!
My * is on a modest machine (Duron 3000+, 512RAM, a good Gigabyte MB and a
cheap PSU). I've got a TDM400P card with one FXS and three FXO. The UPS is
as good as I'm willing to put into the box.
If power fluctuations are known to cause such problems I'll
I'm getting an error back from an AGI Dial command. Weird thing is that it's
STILL performing the Dial.
Here's what I am sending (without the paranthesis):
(EXEC DIAL SIP/1|5|tr)
and here's what I am getting (without the paranthesis):
(510 Invalid or unknown command)
Why would I get this
So, I went ahead and printed to stderr what I was sending to asterisk, and then
I printed to stderr what I get back. You can see it on the Asterisk console of
course. We have the dial command, immediately followed by the 'Invalid
Command', and oh look... Asterisk then goes ahead and dials it
Hello,
I'm running [EMAIL PROTECTED] 2.5
asterisk 1.2.4
zapatel 1.2.2
libpri 1.2.2
on a Dell Poweredge 2850 (1 CPU) with a TE210P
I have 2 t1 circuits using NFAS with dchan on 24 and no backup dchan. I am able
to receive inbound
calls on all channels and can only make outbound calls on
On Thursday 16 February 2006 10:16, Brent Torrenga wrote:
This sounds reasonably plausible. He just might be fooling the busy detect
routine, kinda like how a female voice can trigger DTMF detection.
Ok, but why do you have busydetect turned on? I don't think it's ever done
anything but cause
This is my last update to my issue. Finally my echo problem is
resolved. On Monday morning 2/13/06 I pulled the the zaptel trunk
source. That night after my customers core business hours we built the
new zaptel drivers, rebuilt libpri, asterisk, asterisk-addons. My echo
disappeared almost
FOR THE LIST'S BENEFIT, THIS IS MY EMAIL TO THE LOUD PARTY ON OUR SYSTEM,
THANKS FOR ALL YOUR HELP, HOPEFULLY I HAVE THE ISSUE SOLVED:
Well, I got a series of suggestions as to how to solve your hangup problem.
My favorite suggestion:
LOL... You could try
Hi,
I'm setting up an asterisk server with this hardware configuration:
AMD Athlon 1000 Mhz
256 MB ram
3ware ATA raid controller
2 * Ethernet controller
2 * ISDN HFC controller
One ethernet controller is connected directly to the internet (public
IP)
One ethernet controller is connected to the
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Why do you have immediate set?
*immediate*: Normally (i.e. with immediate set to 'no', the default),
when you lift an FXS handset, the Zaptel driver provides you a
dialtone and listens for digits that you dial, passing them on to
Asterisk. Asterisk
Rich Adamson wrote:
I'm within a couple of weeks of attempting this by interfacing asterisk
with a Siemens CO switch via gr303. All the physical components are in
place, just need to find some time to config asterisk, etc. This CO
switch already has something like 30 remote concentrators (from
Well, I'm about ready to throw Asterisk across the room.
Can someone tell me WHY, when you've sent a Dial command to Asterisk via AGI,
if the callee hangs up the call, Asterisk sends a return code, but if the
caller hangs up, it does not???
This means if an agi script services a call, and
On Thursday 16 February 2006 11:11, Stagg Shelton wrote:
Here are my final configurations
zaptel trunk pulled 2/13/06 approx 10:00am est.
Can you tell us what SVN checkout # and echo canceller you ended up using?
'dmesg' output when you load the module will tell you, as will ztcfg -v.
Also,
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Brent Torrenga
Sent: Thursday, February 16, 2006 11:17 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] RE: Random Disconnects - or ARE they?
FOR THE LIST'S BENEFIT, THIS IS MY
Hi,
Just so I am clear this patch will work with 1.2.4 and requires manual
updating to files and then a recomplie of Asterisk source correct??
Thanks
Ben Klang wrote:
Hello,
I found the same problem very frustrating, mostly because it causes Asterisk
to ignore ACLs and umask settings.
I'm getting the error on the bottom of pages, I'm running this in tandem with 1.4, so not sure if this is an issue, but 1.4 still works (using the same user, password and database as version 2).
Warning: mysql_pconnect(): Access denied for user: '[EMAIL PROTECTED]' (Using password: YES) in
Sean,
I was attempting different settings. I tried immediate=no and yes. Neither
work.
Thanks,
Aldo
On Thu, 16 Feb 2006 11:21:44 -0500, Sean Cook wrote
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Why do you have immediate set?
*immediate*: Normally (i.e. with immediate set to 'no',
After looking at the wakeup-ext.php script there is a lot more I'm
looking for. I'm trying to find something that will let each user
record their own message to be played to them when they get called.
Also when called the user will be required to push say 1 to say ok, and
2 to call back or
Hi,
Could you post the updated patch for 1.2.4
Thanks
Ben Klang wrote:
On Thursday 16 February 2006 11:47, you wrote:
Just so I am clear this patch will work with 1.2.4 and requires manual
updating to files and then a recomplie of Asterisk source correct??
This patch was written
Subject:
[Asterisk-Users] AGI Flakyness *sigh*
From:
Douglas Garstang [EMAIL PROTECTED]
Date:
Thu, 16 Feb 2006 09:24:26 -0700
To:
Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Well, I'm about ready to throw Asterisk across the room.
Can someone
Andrew: Yeah, busydetect=yes == problems. Duly noted!
Alexander:
Sorry, not a PRI line, just a TDM400P.
Would a PRI or BRI not use the D channel to signal busy, anyways? I have a
lot to learn about the workings of ISDN...
One other thing that I did not mention, Are you using a PRI? What are
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
This shouldn't make any difference... check your defines.php and make
sure you have the correct username/password...
define (USER, root);
define (PASS, some_really_strong_secret);
Sean
Joe Pukepail wrote:
I'm getting the error on the bottom of
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
This shouldn't make any difference... check your defines.php and make
sure you have the correct username/password...
define (USER, root);
define (PASS, some_really_strong_secret);
Sean
Joe Pukepail wrote:
I'm getting the error on the bottom of
Are you looking for a custom solution ? We can build
you one.
Regards,
Dovid
--- Michael Sampson [EMAIL PROTECTED] wrote:
After looking at the wakeup-ext.php script there is
a lot more I'm
looking for. I'm trying to find something that will
let each user record
their own message to be
On Thursday 16 February 2006 12:07, Brent Torrenga wrote:
Would a PRI or BRI not use the D channel to signal busy, anyways? I have a
lot to learn about the workings of ISDN...
You'd think so, but some braindead PRI implementations use inband signaling of
call progress, and Asterisk uses inband
Just had Digium take a look at my box:
The following fixed it:
[etc/asterisk/zapata.conf]
trunkgroup=1,24
spanmap = 1,1,0
spanmap = 2,1,2
using logical span 0,2 instead of 1,2 resolved the issue.
Thanks,
Aldo
On Thu, 16 Feb 2006 11:50:39 -0500, Aldo Gonzalez wrote
Sean,
I was attempting
Freddi,
Ok... sure... here's the code. It's about as basic as you can get.
#!/usr/bin/python
import time
import string
import sys
class AGI:
def __init__(self):
self.env = {}
while 1:
line = string.strip(sys.stdin.readline())
if line == '':
I am trying to exactly this using 1.2.4, and it doesn't happne.
DTMF works fine for VM and IVR.
Joe
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On Thursday 16 February 2006 13:18, Douglas Garstang wrote:
As you can quite clearly see, Asterisk sends no return code back to the AGI
script. I really want to understand why this happens. I also don't like the
Are you using AGI() or DeadAGI() ?
until after the call is Hung up. Also, another
Hi, everybody:
I enabled speex in my asterisk box (iax.conf), but no phone call went
throug. At the asterisk console, I used the show modules command and
it did not show the speex codec in the list.
So, I downloaded the speex codec from speex.org, v1.0.5, compiled and
installed in my asterisk
In a dedicated fax server with brooktrout fax cards (analogue), and when
I first setup my * without a UPS. We were noticing that the lines became
un-initialized which required the fax/phone software/drivers to
require re-initialization. On our windows based fax server this required
restarting the
Ok, I got through that error, after recompiling with app_cbmysql asterisk doesn't want to start up. I renamed the app_cbmysql.so file and it came up ok.. Anyone have any advise?
[app_cbmysql.so]Feb 16 13:08:17 WARNING[21558]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/app_cbmysql.so:
Andrew.
Thanks for the reply. I'm using AGI, not DeadAGI. I don't see how I can do post
call processing if Asterisk never returns a return code. My script is blocking,
waiting on input from stdin, which it never gets. Actually, now that I think
about it, what I think is happening, is that when
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
see the previous emails about hand editing the makefile
Sean
Joe Pukepail wrote:
Ok, I got through that error, after recompiling with app_cbmysql
asterisk doesn't want to start up. I renamed the app_cbmysql.so
file and it came up ok..
Did you rebuild asterisk after your speex install?
Mark, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com
Jesus E Zepeda wrote:
Hi, everybody:
I enabled speex in my asterisk box (iax.conf), but no phone call went
throug. At the asterisk console, I used the show modules command and
it did not
I need 10 DID's for it those country's
NicaraguaEl salvadorCosta RicaPanamaHonduras
Thank's
João Carlos MouraNiNeTel
Telecommunications7382 N.W. 35 TerraceMiami, FL 33122 USA
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turby wrote:
is there recomended source files for t.38 pass? latest cvs does not work for me.
is it possible publish working src?
You mean T.38 passthrough? I've just uploaded an asterisk-1.2.4
backport of the lastest svn asterisk/trunk T.38 code to the bugtracker,
and it works swell for
This is my last update to my issue. Finally my echo problem is
resolved. On Monday morning 2/13/06 I pulled the the zaptel trunk
source. That night after my customers core business hours we built the
new zaptel drivers, rebuilt libpri, asterisk, asterisk-addons. My echo
disappeared almost
On Feb 16, 2006, at 1:11 PM, Adolfo R. Brandes wrote:
turby wrote:
is there recomended source files for t.38 pass? latest cvs does not
work for me.
is it possible publish working src?
You mean T.38 passthrough? I've just uploaded an asterisk-1.2.4
backport of the lastest svn
Hello all,
Has anyone figured out a way to send email notifications etc. due to
failed IAX2 registration attempts?
Thanks
-Ron
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Well, I thought and hoped my issue of random hangups on our TDM400P were
related to busydetect=yes in zapata.conf. The behavior of a call being
hungup has not changed, however, since setting the busydetect option to
'no'. Again, the only affected user is my loud talker...
What are some
Hello every one.Please, help.I:1. Run install-pdf from linux to support faxes on my asterisk, according to the instructions at http://www.voip-info.org/wiki/view/Asterisk%40Home+Handbook+Wiki+Chapter+8 2. Made the configurations throuhg AMP in a.Setup-Inbound Routing-(the only route
On 2/16/06, Brent Torrenga [EMAIL PROTECTED] wrote:
Well, I thought and hoped my issue of random hangups on our TDM400P were
related to busydetect=yes in zapata.conf. The behavior of a call being
hungup has not changed, however, since setting the busydetect option to
'no'. Again, the only
Maybe people are just hanging up on him cause he is talking to loud.
Doug Lytle wrote:
Brent Torrenga wrote:
I have one use on our PBX who has been experiencing seemingly random
disconnects. The user is on the same LAN as everyone else, using the
same
Brent,
The last time I was
Having followed this thread, it seems to me that the simplest way to test
for the busydetect hangup is to get the guy to make a few phone calls on
someone else's phone. If the hangup happens, then it's the guy. If it
doesn't, it's his station.
-Original Message-
From: Michael Sampson
Colin Anderson wrote:
Having followed this thread, it seems to me that the simplest way to test
for the busydetect hangup is to get the guy to make a few phone calls on
someone else's phone. If the hangup happens, then it's the guy. If it
doesn't, it's his station.
The way that I ran
I just tried to go from CAS to PRI on my T1 (Sangoma), and failed pretty
badly. Seemingly everything worked -- Asterisk would see the incoming
call (including CID and DID info), try to route it, and fail -- giving
me a telco (not Asterisk) call failure message. My zapata.conf and
zaptel.conf
CLI show channels
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of jonny hashem
Sent: Friday, 17 February 2006 12:46 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] show calls
HI:
what is command on console to know how many calls are
hi,
My question is may be a bit out of scope but I don't know where to turn,
I have a 1760 with a ccme 24 user licences 1 bri card.
I want to configure a bri card in a cisco 1760 on port 0/0,
the card is new, seen by the router, show isdn status gives layer 1 desactived , layer not activated,
You need to recompile Asterisk itself after installing Speex. Do a make
clean, make, make install. I usually stop asterisk before that last step, by
the way!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jesus E Zepeda
Sent: Friday, 17 February 2006
Wrong list. You
want asterisk-biz.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of JOAO CARLOS MOURA
Sent: Friday, 17 February 2006
9:06 AM
To:
asterisk-users@lists.digium.com
Subject: [Asterisk-Users] DID's
I need 10 DID's for it those country's
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