This feature has worked for us since ver 1.0 (not cvs)
Alvaro Parres wrote:
Josheph:
I had have that problem, and it get solve when i take out the
incominglimit from my sip.cfg
Also if you send you sip.cfg and extensions.cfg will be easier to
help you
Tray it.
Alvaro Parres
I also have no trouble on production systems 2.6.9/10 Gentoo-dev-sources
On Sun, 2005-01-16 at 15:14 +1300, Matt Riddell wrote:
Brian West wrote:
I have never had an issue with 2.6.9 with asterisk.
I second that.
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as the subject states I have a TDM400 that when a call is answered
asterisk runs the dialplan even if i hang up it NEVER detects the hangup
and I am also having a hard time with CID info I don't get that either.
most of our production machines are PRI based and I have little
experience with the
(${GENERALMAIL})
On Sat, 2005-01-15 at 18:29 +1300, Matt Riddell wrote:
Michael George wrote:
On Tue, Jan 18, 2005 at 04:16:08AM -0600, Justin Carlson wrote:
no i was using line 1 for testing /w fxs module and i never changed it
back
Also, could you show us the contents of your [routing] context
I need to know if this works and if so does anyone have a sipura config
to post? I have looked and not found anything conclusive.
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To
is the hint
99,hint,ZAP/1
supposed to work or how do I get the lights on the phones to display
channels in use in addition to extensions in use?
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no we have a tdm400 at this site does this still apply?
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no i was using line 1 for testing /w fxs module and i never changed it
back
On Fri, 2005-01-14 at 07:43 -0500, Michael George wrote:
On Mon, Jan 17, 2005 at 08:12:24AM -0600, Justin Carlson wrote:
hi all,
We have a TDM400 card with 4 wfo modules. now the modules load fine
and when
hi all,
We have a TDM400 card with 4 wfo modules. now the modules load fine
and when i start asterisk with on phone line connected it just starts
spewing these messages:
-- Starting simple switch on 'Zap/4-1'
Jan 13 12:59:47 NOTICE[10831]: chan_zap.c:5367 ss_thread: Got event 2
what is wrong with running asterisk with the -pg flags at startup?
On Mon, 2005-01-03 at 19:13 +0200, Gilad Ben-Yossef wrote:
Matt Schulte wrote:
Had a good question for the list, it seems whenever I work in an
Asterisk console or on the machine normally I get jitters on any audio
going
double post
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what was wrong with logrotate?
On Thu, 2004-12-30 at 10:57 -0500, Matt Gibson wrote:
Hi Randy,
Randy MacKay wrote:
I do about 500 calls per day on average volume and about 750 on heavy
volume and find it necessary to run a logger rotate every other day...
other then that I can go on for a
Toggle the break key in the web config on your snom and then the
break/transfer key will actually be the transfer key.
On Sun, 2004-12-26 at 09:09 -0500, steve szmidt wrote:
On Tuesday 21 December 2004 10:36 pm, Tracy R Reed wrote:
I am having a hell of a time with transfers.
First the
you could comment that portion out and rebuild?
On Fri, 2004-12-17 at 13:15 +0100, Alessio Focardi wrote:
Hi,
since I run asterisk as root with a CLI open on TTY12 I was wondering
if the ! (shell) command can be disabled from the config, for safety
reasons it seems me usefully.
Tnx for
we also have observed that you must reboot the snom phones EVERY time
you reload the dial plan or restart the server we are using *Asterisk
1.0.0
On Fri, 2004-12-17 at 21:47 +0100, Joris Trooster / Interstroom wrote:
In your extensions.conf create a hint:
exten = 215,hint,SIP/215
On the
I wonder if you can put can reinvite=yes in the iax2.conf file like we
use in our sip.conf file to do what you are requesting.
I believe it should tell the phones to do what you wish
On Thu, 2004-12-16 at 17:40 +0200, CuPoTKa wrote:
Hello!
I have a number of IAX clients behind a NAT (on the
Thank you! I will try again tomarow
On Tue, 2004-12-14 at 14:05 -0500, Leif Madsen wrote:
On Tue, 14 Dec 2004 03:22:47 -0600, Justin Carlson [EMAIL PROTECTED] wrote:
If anyone has a working unidencomm.txt and unidenMACOFPHONE.txt file Could
you please post it.
Hi Justin,
I am using
we also use the 220 but with the additional button panel with great
results!
On Tue, 2004-12-14 at 07:25 -0600, Gerald J. Puhl wrote:
Does this phone have LEDs showing lines in-use?
Thanx!
Gary P.
Tracy R Reed wrote:
On Mon, Dec 13, 2004 at 12:50:54PM -0600, Gerald J. Puhl spake thusly:
Hello all,
I have a uip200 for testing and I can't seem to get the phone to
register to my * server. I have configured the unidencomm.txt and the
unidenMACOFPHONE.txt files and the phone tries to register but * comes
back with a 403 Forbidden message in sip debug, the phone simply
make sure that the extension dialed has the ability to transfer calls
and that the break key function is set correctly in the snom web admin.
otherwise when you hit the transfer key it simply drops the call
On Tue, 2004-12-07 at 18:44 +, Asterisk wrote
:
I cannot get the transfer button
Doesn't the IAXy device have a power adjustment like the sipuras?
On Thu, 2004-08-05 at 18:16, Glen Hinkle wrote:
For anyone interested, the banshee screen I was experiencing was due to
my cordless phone. I used a normal corded phone without separate power
it was fine.
I suppose there
Has anyone integrated asterisk with current version of rt. I followed
the Wiki but I only get as far as hold on while i create a ticket then
it hangs up. I don't see it connect to the rt-soap-server.pl script
running on the console of my rt machine. any help would be greatly
appreciated.
We use an IAX2 trunk to our remote office and would like for the
receptionist to be able to transfer incoming calls from this trunk. but
all calls come in as one user, Is there a way to get a breakout on the
flash GUI of the incoming calls?
Thanks,
Justin
Thank you for the prompt reply but when I add 7;8;9, in my button number
field the iax2 button goes away. i just got .10 today
.
On Mon, 2004-06-28 at 11:51, Nicolas Gudino wrote:
Hi Justin,
Justin Carlson wrote:
We use an IAX2 trunk to our remote office and would like
I have had similar troubles and doing a modprobe -r zaptel then
re-loading the zaptel modules seems to cure it. ( if you unload the
wcfxo ztdummy wct1xxp etc it leaves the zaptel module loaded.)
On Tue, 2004-06-08 at 07:25, Rich Adamson wrote:
I've been playing with two pieces of hardware:
have you tried commenting out the dtmf lines in your sip.conf we had
similar problems with our snom 200's and after commenting out the dtmf
lines in sip.conf asterisk reload they worked great :-)
On Wed, 2004-06-02 at 11:36, Lee Norvall wrote:
Hi
I have 2 x SIP hand phones. I have set
we have the same problem could you please send me the chan_sip2 info.
Thanks!
On Sat, 2004-04-24 at 14:23, Geert Nijpels wrote:
Ian White wrote:
On Apr 22, 2004, at 23:48, Olle E. Johansson wrote:
Geert Nijpels wrote:
Ian White wrote:
On recent releases of the snom200
where can I get the 2.05 firmware all i see is the 2.04 firmwares :-)
also anyone got a fix for the horrible speaker phone on the 200's
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I don't think your DTMF is set right look in sip.conf for the dtmf
directive for your phones.
cheers!
On Tue, 2004-05-04 at 13:41, Michael Picher wrote:
Searched the archives thoroughly...
Can't find this specific problem...
Simple setup with Asterisk on RedHat. No voice cards in the
we are getting these errors too which cvs was it fixed in ? we just
upgraded to cvs-stable from friday to see if that would help.
On Sun, 2004-05-02 at 21:45, brian k. west wrote:
I think this was fixed in CVS-HEAD because I do not see that message
in the src at all while looking to see if t
a warning.
bkw
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Justin Carlson
Sent: Monday, May 03, 2004 3:21 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] IAX2
we are getting these errors too which cvs was it fixed
is there a firmware for IAX for the snom 200's. or are there any other hard
phones that use iax(2)?
Thanks in advance!
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We also are having randomly dropped calls with our IAX2 connections, we
have tried IAX2 with and without trunking enabled. the phones are snom
200's with SIP and there is an asterisk box at each site so no sip nat
problems.
-Original Message-
From: [EMAIL PROTECTED]
or
through say oneunified)
If you figure this out, please let us know here. I'm pretty much at a
loss as to what could be causing it.
Justin Carlson wrote:
Hi all,
We keep getting these and all the calls between these two asterisk boxes
get
dropped. what is going on here, I
Hi all,
We keep getting these and all the calls between these two asterisk boxes get
dropped. what is going on here, I have been trying to solve this problem on
my own but maybe I don't have the trunk setup right. also I have posed the
output of my full log of the machine with the zap
dido
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of
[EMAIL PROTECTED]
Sent: Wednesday, April 07, 2004 2:41 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Problems with IAX2?
Hi
I am also having jitter trouble on IAX2, and I can vouch
that the
I need to be able to use a variable that has the calling extension number
rather than the called.
thanks.
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:
[EMAIL PROTECTED]Subject: RE: [Asterisk-Users] Need a
list of asterisk built-in variables
The ${CALLERIDNUM} variable has the calling extens
number.
Regards, Pedro Goncalves
-Original Message- From:
Justin Carlson [mailto:[EMAIL PROTECTED]]
Sent: terça-feira, 6 de
I have a suse 8.2 installation of mpg123 and I have no problems with the id3
tags
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Philipp von
Klitzing
Sent: Tuesday, April 06, 2004 11:37 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] mpg123 issue and
]
[mailto:[EMAIL PROTECTED] Behalf Of Olle E.
Johansson
Sent: Tuesday, April 06, 2004 12:12 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Need a list of asterisk built-in variables
Justin Carlson wrote:
I need to be able to use a variable that has the calling extension number
rather than
] Need a
list of asterisk built-in variables
Suppose EXT1 makes
call to EXT2. Then the ${CALLERIDNUM} is the number of EXT1 while ${EXT} is
the number of EXT2.
Any
doubts?
Regards,
Pedro
Goncalves
From: Justin
Carlson [mailto:[EMAIL PROTECTED] Sent
if you don't give them the pass code they can't hang-up or transfer calls
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Adams, Gavin
Sent: Friday, April 02, 2004 7:30 AM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] ANNOUNCE: Flash Operator Panel
this is not where to send your unsubscribe to
!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Altus Snyman
Sent: Friday, April 02, 2004 7:20 AM
To: asterisk
Subject: [Asterisk-Users] UNSUBSCRIBE
just type it in it will remain until you restart your browser. ( it does
not disappear and you do not have to hit enter or anything like that)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Brian Capouch
Sent: Friday, April 02, 2004 3:45 AM
To: [EMAIL
we also would require more buttons, at least 40, can we get a multipage
view. right know I run multiple servers on the same page to get the effect
of having 3 pages.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Nicolas
Gudino
Sent: Friday, April 02, 2004
vmail.cgi seems to be written in perl so modifying it should require
knowledge of perl and vi
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Chris Clifton
Sent: Friday, April 02, 2004 10:51 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] voicemail
How
Hi all,
I have a problem with echo and silence in the middle of calls. the echo
problem is that in the first 5 to 10 seconds of a call there is echo on the
sip side but not on the PSTN side, also the echo will randomly come back in
the call sometimes, I'd say 3 out of 10 calls. the
our cvs is 02/25/04
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Juan J.
Sierralta P.
Sent: Thursday, April 01, 2004 11:56 AM
To: Asterisk Users
Subject: Re: [Asterisk-Users] Zap Channels Hang
On Thu, 2004-04-01 at 10:37, Sergi Gabunia wrote:
Hi,
I
] Echo's and dropped calls
Do you have
echocancel=yes
echocancelwhenbridged=yes
echotraining=yes
In your zapata.conf file? Wiki is good for this -
John V.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Justin Carlson
Sent: Thursday, April 01, 2004 9:45 AM
volume -
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Justin Carlson
Sent: Thursday, April 01, 2004 10:21 AM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Echo's and dropped calls
on both the box with the zap interface and the remote office
This I one of the things we have been looking for!!! I just installed it in
about 5mins and works great!!!. Excellent work.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Nicolas
Gudino
Sent: Thursday, April 01, 2004 2:52 PM
To: [EMAIL PROTECTED]
Subject:
you probably need to add a correct host entry in your /etc/hosts file for
your machine it goes
ip namealias
192.168.1.1 asterisk.goober.org asterisk
so
192.168.1.1 asteriskasterisk.googber.org
is
I am sorry if this is a silly question but I can not seem to locate the
festival binaries. does this come with asterisk or is it another project?
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a google search before
asking...
google 'festival asterisk install'
http://www.voip-info.org/tiki-index.php?page=Asterisk+festival+installation
-Heison
On Fri, Mar 19, 2004 at 04:10:46PM -0600, Justin Carlson wrote:
I am sorry if this is a silly question but I can not seem to locate
do you know if there is a way to get the conference button on the snom 200's
to work?
-Original Message-
From: Greg Retkowski [mailto:[EMAIL PROTECTED]
Sent: Monday, March 15, 2004 1:31 PM
To: [EMAIL PROTECTED]; Justin Carlson
Subject: Re: [Asterisk-Users] Conference call?
On Mon, 15
.
-- Greg
Greg Retkowski / I.T. Infrastructure Consultant /)/|//`
[EMAIL PROTECTED] http://www.rage.net/~greg/ C:408-455-3913 /|/ /_/
On Mon, 15 Mar 2004, Justin Carlson wrote:
do you know if there is a way to get the conference button on the snom
200's to work?
-Original Message
: [Asterisk-Users] Night menu not working
These suggestions may not help get your daytime stuff working, but it
should make life easier later.
On Thu, 2004-03-11 at 14:53, Justin Carlson wrote:
[general]
static=yes
writeprotect=no
[globals]
MARYKAY = 21
RECEPTIONIST = 20
KATHY = 22
[daytime
Hi all,
I am trying to get day and nighttime menus to work in * and no matter what
time I specify the first include entry that matches the number dialed is
used. I have included my extentions.conf and my sip phones have a default
context of default.
[general]
static=yes
writeprotect=no
yes be sure you are using ULAW and I found that 9600 was the baud rate to go
with. 14400 seemed to be unreliable.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Mark
Messmore, Technical Support, University Telcom Inc.
Sent: Friday, March 05, 2004 2:15 PM
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