Hm, I have installed ring group and I'm using it, when I want that all phones in ringall group would ring. Maybe I must create memoryhunt or hunt group. And what about Group Number.
2006/7/28, Tim P [EMAIL PROTECTED]:
You could setup a ring group that included all extensions in your
inbound route,
On Fri, Jul 21, 2006 at 06:13:50PM -0500, brandon kruz wrote:
in addition to russel
use
(in ubuntu)
sudo netstat
or man netstat for further, more precise methods
look for your specific port
eg
sudo netstat -a | grep 5060
and it shoudl tell you the process name, and what directory it is
On Mon, Jul 24, 2006 at 10:20:48PM +1000, Devraj Mukherjee wrote:
Hi Alejandro,
Thanks for your suggestions. Where did you fetch your rpms?
I had to fix up the init scripts for everything to work
Which init script? For which distribution?
What exactly were your fixes?
What is the number
On Mon, Jul 24, 2006 at 05:09:52PM +1000, Devraj Mukherjee wrote:
Hi Everyone,
I was having a lot of trouble starting up Asterisk and zaptel using
the init.d scripts. I have worked on the scripts and now the zaptel
script so it reads preferences of /etc/sysconfig/zaptel file and
starts the
On Fri, Jul 28, 2006 at 04:50:50PM +0200, Tijl Van den Broeck wrote:
I installed the following packages as well:
ii libzap-dev 1.0.1-1 Zapata
telephony interface library (developm
ii libzap11.0.1-1 Zapata
On 07/29/06 02:49 Miles Scruggs said the following:
http://forum.4psa.com/showthread.php?t=455
Take it for a ride around the block and tell them what you think. As
powerful as the config files, and command line interface is, there is
is there anywhere we can take a look at screenshots
I tried to edit the cdr import function but I didn't know where it placed or
what function to edit ,
Please can you tell me where to place this function
exten = s,1,Set(CDR(userfield)=${SIPCHANINFO(recvip)})
to have it stored in the mysql record .
I am using [EMAIL PROTECTED] 2.6
Regards
Hi Martin,
No exactly. The Fritz!Box is connected to Asterisk using SIP. Not a direct
connection between FXS ports and Asterisk.
I would like to use this box like a Sipura 3000. This Sipura has 1 FXO port
and 1 FXS port. You can use and register these ports in Asterisk
independently. You
On Fri, 2006-07-28 at 15:24 -0500, Rich Adamson wrote:
If the source is not installed by default, is that not a violation of
the GPL license?
No, because it is available for download. If not my Linksys router would
have to be twice as big, I can download the necessary files from their
site.
Hello there all, i am using an agi python script. It is kinda from an example in the ATOF book ( O'Reilly)it simply answers the phone, receives 5 DTMF digits, and writes those digits to a text file.
however, it isn't working.The script is in python, and i have stderr writing out some debug
once more :-)
hi,
I see. No, that will not work with this box and the original firmware. :-(
You could send me the pages and descriptions you found on manipulated
firmwares for use with asterisk off this list. Then I can take a look at
them and tell you, if it will work or what it will do. :-)
Hi
On 7/27/06, Luki [EMAIL PROTECTED] wrote:
There is this old patch that does remote MWI over IAX (among other
things). I used it on earlier versions and it worked quite nicely.
This was before 1.2 so it may no longer work at all. At the very least
it will likely required some updating.
On Fri, Jul 28, 2006 at 04:08:19PM -0500, shawn bright wrote:
i would use a dial plan, but we are monitoring about 1200 units in the
field, i thought a dial plan would be a little long or complex for that. I
suppose that i could use a dial plan and set guys up by editing the
extensions.conf
On Thu, Jul 20, 2006 at 10:53:26PM -0400, augustynr wrote:
Hi,
I got realy tired of looking at Asterisk lists in Outlook so I
moved it into the phpBB2 type forum. It seems to be working well
for me and I think some of you may find it usefull too.
So here it is at:
Stop asterisk and run it from the command line directly(asterisk -gc).
For some reason AGI scripts only output to the original Asterisk
session, not remotely connected Asterisk sessions(asterisk -r)
MATT---
On 7/29/06, shawn bright [EMAIL PROTECTED] wrote:
Hello there all,
i am using
Yep, that worked, thanks a lot. Now i can at least see whats going wrong.thanks again.-shawnOn 7/29/06, Matt Florell
[EMAIL PROTECTED] wrote:Stop asterisk and run it from the command line directly(asterisk -gc).
For some reason AGI scripts only output to the original Asterisksession, not
There currently exist no such option. But
you are free to try to add it.
SNIP
___
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To UNSUBSCRIBE or update options visit:
Does anyone know how to switch out the background image? I
cannot find it defined anywhere.___
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On Sun, Jul 23, 2006 at 12:27:43PM -0500, Russell Bryant wrote:
- [EMAIL PROTECTED] wrote:
Im trying to install asterisk 1.2.10 on a new debian 3.1r2 machine
and every
time i try to make it i get an
Configure: error: termcap support not found
Make: *** [editline/libedit.a]
If you put an image named background.jpg in the folder with panel it will be put behind the flash file.On 7/29/06, Jordan Novak
[EMAIL PROTECTED] wrote:
Does anyone know how to switch out the background image? I
cannot find it defined anywhere.
On Mon, Jul 24, 2006 at 07:33:38AM -0700, Ryder Brook wrote:
I have an AAH, seems to be Asterisk version 1.2.7.1.
It seems to be rebooting everyday around 8:30 am and the office goes hay
wire, as this is a doctor's office, even if it's for a brief minute. Nothing
remarkable in the logs.
Maybe the question is, how can I call someone right after I something
happens, in this particular case if the Dial is not answered.
Manrique Feoli escribió:
Hi all,
I am receiving a call on one E1 and try to set up a call on another
E1, if the second call succeds, fine but if
On Tue, Jul 25, 2006 at 03:32:20PM +1000, Nick Hoffman wrote:
Hi guys. I just stumbled upon
http://www.voip-info.org/wiki/index.php?page=Asterisk+G.729+Licensing and
read the section titled Warning. I'm a bit confused now. Are you
violating the GPL (or any other license) if you sell a
How do you recompile individual source modules?
I need to make a small change (addition) to chan_zap.c. I read somewhere
you can recompile individual module source without the need to recompile
the entire asterisk sources each time at change is made. Can someone
tell this 'C' noob how to do
On Tue, Jul 25, 2006 at 08:12:07PM +1000, Eric Bishop wrote:
Anyone know if it possible to create binary/obfuscated/ human unreadable
extensions.conf/sip.conf etc.? We would like to deploy a system in an
environment where not giving out root is still not enough. We want to hide
the contents of
Did you look on the site?http://www.4psa.com/products/voipnow/demo.phpOn 7/29/06, Dinesh Nair
[EMAIL PROTECTED] wrote:
On 07/29/06 02:49 Miles Scruggs said the following: http://forum.4psa.com/showthread.php?t=455 Take it for a ride around the block and tell them what you think.As
powerful as
On Tue, Jul 25, 2006 at 10:21:16AM -0500, Carlos Chavez wrote:
On Tue, 2006-07-25 at 20:12 +1000, Eric Bishop wrote:
Anyone know if it possible to create binary/obfuscated/ human
unreadable extensions.conf/sip.conf etc.? We would like to deploy a
system in an environment where not giving
- Bart Fisher [EMAIL PROTECTED] wrote:
I need to make a small change (addition) to chan_zap.c. I read
somewhere
you can recompile individual module source without the need to
recompile
the entire asterisk sources each time at change is made. Can someone
tell this 'C' noob how to do
- Simon Austin [EMAIL PROTECTED] wrote:
I have confirmed that GET VARIABLE doesn't return global variables in
version 1.2.10 and submitted the following bug report:
http://bugs.digium.com/view.php?id=7609
I'm not sure if you have seen it, but I posted a patch to your bug report about
an
- Douglas Garstang [EMAIL PROTECTED] wrote:
context new_pbx_betty_start {
_X. = {
for (x=0; ${x} 3; x=${x} + 1) {
Verbose(x is ${x} !);
}
};
}
Here's the output.
The var x never gets incremented! Is this a bug?
The
Hello :-)
I just read, that the agentcallbacklogin will be marked as depreciated in
v1.4 and we should use dynamic members.
What do you think of this?
Is it possible to use the dynamic members instead with all the features?
1. Maybe somebody can give me a hint, how to set up the following with
context new_pbx_betty_start {
_X. = {
for (x=0; ${x} 3; x=${x} + 1) {
Verbose(x is ${x} !);
}
};
}
I would have to see the output of show dialplan new_pbx_betty_start to
know exactly what is going on. However, I'm guessing that
If I understand, I cd to asterisk source folder and run make - it take
card of rest?
Also, when/why should you use astxs?
Bart
Russell Bryant wrote:
- Bart Fisher [EMAIL PROTECTED] wrote:
I need to make a small change (addition) to chan_zap.c. I read
somewhere
you can recompile
I can retrieve GLOBAL variables that I set in
AGI...I never tried setting them in extensions.conf and then retrieving...but I
would have assumed the same result...but you never know I guess
- Original Message -
From:
Simon
Austin
To: Asterisk Users Mailing List -
On Thu, Jul 27, 2006 at 09:04:24AM +0200, [EMAIL PROTECTED] wrote:
Hello,
is it possible to restart the wct4xxp kernel module and start again
without stopping Asterisk?
In trunk (using 'zap restart': bug #6255)
i tried to unload chan_zap.so but rmmod says the module is in use.
In
Bart Fisher [EMAIL PROTECTED] wrote:
If I understand, I cd to asterisk source folder and run make - it take
card of rest?
Also, when/why should you use astxs?
I repeat in other words what Russell said:
When you have a clean source tree and type make a lot of source
files are compiled. When
Russel, I did see your note. Thanks for the patch. I haven't had a
chance to apply it yet. I hope to apply it tommorow. I'll let you
know the results as soon as possible.
Thanks for your quick response. That was the fastest response to a bug fix request I've ever seen.
Cheers,
- SimonOn
Hello,
I recently updated some Polycom 501 phones to the new 1.6.7
firmware, and have lost the ability to do One Touch voicemail access via
the messages button.
I've verified that I have the correct XML tags set in the phone config,
I.E.:
msg.bypassInstantMessage=1
mwi
I actually did get it to work, by removing _all_ spaces from the for line...
for (x=0;${x}3;x=${x}+1) {
This works for me. It's just a matter of finding WHICH space is breaking it.
-Original Message-
From: Rushowr [mailto:[EMAIL PROTECTED]
Sent: Sat 7/29/2006
Douglas,
Awesome! I don't know why I didn't get to the point of removing all the
spaces, probably got distracted by some shiny object ;-)
Anyway, thanks for the update!
Rushowr
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Douglas
Garstang
Sent:
How about up.oneTouchVoiceMail=1 in your sip.cfg
Peter
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Greg Boehnlein
Sent: Sunday, 30 July 2006 8:37 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Polycom 1.6.7 Firmware Messages Button
Hi,I am new to the list and in need of help. I have asterisk 1.2.10 setup and configured to receive did on iax2 channel. All was working fine till this evening after update of dialplan. No I get the following and no audio when with incoming calls - chan_iax2.c:6756 socket_read: Ooh, voice
Hello,
Ive got asterisk running and almost working
with Panasonic KX-TD1232
I said almost, because theres a strange behaviour
when I make calls.
---
-
-
---
| SIP | -- | ASTERISK | -- | PANASONIC
| | PSTN
Geez. This is starting to sound like Microsoft licensing.
On 7/29/06, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Tue, Jul 25, 2006 at 03:32:20PM +1000, Nick Hoffman wrote: Hi guys. I just stumbled upon
http://www.voip-info.org/wiki/index.php?page=Asterisk+G.729+Licensing and read the section
On Sun, 30 Jul 2006, Peter Johnson wrote:
How about up.oneTouchVoiceMail=1 in your sip.cfg
Peter
Ahhh... that tag wasn't in my config generator script, so I must have set
it by hand in the old ones. That does the trick!
I owe you a beer!
--
Vice President of N2Net, a New Age
Hello Pablo, I think you should decribe with
details how are you routing the call between the SIP device and the
extensions.
Pablo
- Original Message -
From:
Pablo Mora
To: asterisk-users@lists.digium.com
Sent: Thursday, July 27, 2006 10:18
PM
Subject:
How is asterisk connected to the Panasonic KX-TD1232?
On 7/27/06, Pablo Mora [EMAIL PROTECTED] wrote:
Hello,
I've got asterisk running and almost working with Panasonic KX-TD1232
I said almost, because there's a strange behaviour when I make calls.
---
On Fri, Jul 28, 2006 at 02:33:11PM +0100, Kenny Millington wrote:
Koen Van Impe wrote:
I use logrotate too, because I didn't know of the functionality in Asterisk.
Logrotate works fine for me though.
Ok, I believe I see the problem here!
I was told (apparently erroneously) that asterisk
You have a config generator script for the Polycom XML files? What did you
build that with?
-Original Message-
From: Greg Boehnlein [mailto:[EMAIL PROTECTED]
Sent: Sat 7/29/2006 7:41 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Tom Vile wrote:
Did you look on the site?
http://www.4psa.com/products/voipnow/demo.php
Man that looks nice. Kinda reminds me of the Plesk.
Anyway, I've put up a screenshot with the original post at:
http://www.sineapps.com/news.php?rssid=1399
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Tzafrir Cohen wrote:
On Thu, Jul 20, 2006 at 10:53:26PM -0400, augustynr wrote:
Hi,
I got realy tired of looking at Asterisk lists in Outlook so I
moved it into the phpBB2 type forum. It seems to be working well
for me and I think some of you
On Fri, Jul 28, 2006 at 02:04:10PM +0200, Administrator TOOTAI wrote:
Morning everybody,
I try to install an asterisk test server with trunk branch and get this
error when compiling zaptel. Asterisk core compile fine as well as SVN
1.2 branch. It's a Debian SARGE running on 2.4.27 kernel.
On Sat, Jul 29, 2006 at 09:05:42AM -0500, Matt Florell wrote:
Stop asterisk and run it from the command line directly(asterisk
-gc).
For some reason AGI scripts only output to the original Asterisk
session, not remotely connected Asterisk sessions(asterisk -r)
Only to the first
On Sun, Jul 30, 2006 at 03:33:49PM +1200, Matt Riddell (NZ) wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Tzafrir Cohen wrote:
On Thu, Jul 20, 2006 at 10:53:26PM -0400, augustynr wrote:
Hi,
I got realy tired of looking at Asterisk lists in Outlook so I
moved it into the phpBB2
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