On Jul 7, 2006, at 11:15 AM, Cory Andrews wrote:
x-tad-smallerFor 12 households, you could probably get a business class DSL or Cable broadband internet connection and use it for Voice. Then get something like SIP trunking in place and maintain a few analog POTS lines for local calls and 911
There does not seem to be any make uninstall for Asterisk 1.2.9.1 and
Zaptel 1.2.6...
I tried to apply an uninstall patch but got many Hunk errors from both
1.2.9.1 and latest SVN:
http://bugs.digium.com/file_download.php?file_id=8805type=bug
Is there a reason that there is no make uninstall?
On Sat, Jul 08, 2006 at 04:46:00PM +1000, Carey O'Shea wrote:
There does not seem to be any make uninstall for Asterisk 1.2.9.1 and
Zaptel 1.2.6...
I tried to apply an uninstall patch but got many Hunk errors from both
1.2.9.1 and latest SVN:
On Sat, 2006-07-08 at 10:22 +0300, Tzafrir Cohen wrote:
On Sat, Jul 08, 2006 at 04:46:00PM +1000, Carey O'Shea wrote:
There does not seem to be any make uninstall for Asterisk 1.2.9.1 and
Zaptel 1.2.6...
I tried to apply an uninstall patch but got many Hunk errors from both
1.2.9.1
On Sat, Jul 08, 2006 at 05:43:41PM +1000, Carey O'Shea wrote:
On Sat, 2006-07-08 at 10:22 +0300, Tzafrir Cohen wrote:
On Sat, Jul 08, 2006 at 04:46:00PM +1000, Carey O'Shea wrote:
There does not seem to be any make uninstall for Asterisk 1.2.9.1 and
Zaptel 1.2.6...
I tried to
will you care to sell them? how much?
they work for me ;)
On 7/6/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
I could have told you that. I have 4 handy tones wasting in my basement.
-- Original message --
From: calvis [EMAIL PROTECTED]
Polycom 501
Grandstreams
Hi,
Does anybody know if it is possible to set the outgoing MSN to a different
value than the default set in misdn.conf for a single call only via chan_misdn
0.3.x, and if so, how to do it? I can't find any info on how to do this via
Google, and I've tried a few things myself, none of which seem
On Sat, 2006-07-08 at 10:58 +0300, Tzafrir Cohen wrote:
On Sat, Jul 08, 2006 at 05:43:41PM +1000, Carey O'Shea wrote:
On Sat, 2006-07-08 at 10:22 +0300, Tzafrir Cohen wrote:
On Sat, Jul 08, 2006 at 04:46:00PM +1000, Carey O'Shea wrote:
There does not seem to be any make uninstall for
Hello
I had an Asterisk installation working fine for CallerID on BT analog lines
using a Digium analog 4 port card. However, user switched to TalkTalk
without telling me and CallerID no longer works. However, if you connect a
UK CallerID capable phone into one of these analog lines directly
On Sat, Jul 08, 2006 at 07:27:37PM +1000, Carey O'Shea wrote:
On Sat, 2006-07-08 at 10:58 +0300, Tzafrir Cohen wrote:
On Sat, Jul 08, 2006 at 05:43:41PM +1000, Carey O'Shea wrote:
On Sat, 2006-07-08 at 10:22 +0300, Tzafrir Cohen wrote:
On Sat, Jul 08, 2006 at 04:46:00PM +1000, Carey
On Sat, 2006-07-08 at 13:00 +0300, Tzafrir Cohen wrote:
On Sat, Jul 08, 2006 at 07:27:37PM +1000, Carey O'Shea wrote:
On Sat, 2006-07-08 at 10:58 +0300, Tzafrir Cohen wrote:
On Sat, Jul 08, 2006 at 05:43:41PM +1000, Carey O'Shea wrote:
On Sat, 2006-07-08 at 10:22 +0300, Tzafrir Cohen
2006/7/7, Florian Overkamp [EMAIL PROTECTED]:
Olivier wrote: Do you need to buy an SIP/MGCP spare licence (GPL-SW-SM-UL-7960=) along a Cisco 7960 hardphone (GPL-CP-7960G=) when you simply want to connect it to a SIP enabled Asterisk server ?
Yes, as far as our sales rep can tell us.How far shall
Salve Tzafrir, *!
Thank you for your kindly (and good) support. :)
I found a solution for me to get a colored CLI on a
vserver, but to publish it for other *-users,
I feel it must be smarter ;)
On Tue, 04 Jul 2006, Tzafrir Cohen wrote:
safe_asterisk has a flawed logic: it assumes that the tty
On 20:39, Fri 07 Jul 06, Mike Dent wrote:
Could you pleae explain a little more how this works with Asterisk?
For sip:
Add a line like this in your dialplan for every phone you
want to monitor:
exten = 6000,hint,SIP/6000
Then in the tftp config file for the phone add speeddials
for the 6000
2006/7/8, Olivier Krief [EMAIL PROTECTED]:
2006/7/7, Technical Support [EMAIL PROTECTED]:
We use NV's fax detection and it works very well.
(However this can still congest your system with junk
faxes).
Thanks for the tipRegards
___
--Bandwidth
On 12:33, Sat 08 Jul 06, Olivier wrote:
2006/7/7, Florian Overkamp [EMAIL PROTECTED]:
Olivier wrote:
Do you need to buy an SIP/MGCP spare licence (GPL-SW-SM-UL-7960=)
along a Cisco 7960 hardphone (GPL-CP-7960G=) when you simply want to
connect it to a SIP enabled Asterisk server ?
On Fri, 2006-07-07 at 15:09 -0400, Doug Lytle wrote:
Anthony Davis wrote:
We updated our systems to 1.2.9.1 (from 1.2.4) about 3 weeks ago.
It was supposed to be fixed in 1.2.9.1
A quick check on the server shows that the .WAV and .GSM files were
deleted, but that the
On Sat, Jul 08, 2006 at 10:59:49AM +0100, Angus Comber wrote:
I had an Asterisk installation working fine for CallerID on BT analog lines
using a Digium analog 4 port card. However, user switched to TalkTalk
without telling me and CallerID no longer works. However, if you connect a
UK
On Sat, Jul 08, 2006 at 12:34:42PM +0200, Robert Michel wrote:
Salve Tzafrir, *!
Thank you for your kindly (and good) support. :)
I found a solution for me to get a colored CLI on a
vserver, but to publish it for other *-users,
I feel it must be smarter ;)
On Tue, 04 Jul 2006, Tzafrir
On Sat, Jul 08, 2006 at 08:16:11PM +1000, Carey O'Shea wrote:
I'm not
complaining, I'm just saying that removing things manually means that
you are likely to miss things.
And I'm saying that when removing things automatically you're also
likely to miss things. Or delete to much. Or both. If
hi,
in your init-misdn.conf (or misdn.conf, not sure now...) you can
choose the MSNs for your incoming Ports or Outgoing ports,
msns=3223242,3223243,3223244
for example.
Then in your calls, just set the outgoing callerid for your trunk, to
one of them. Be aware that as far as i know you must
Steve Kennedy wrote:
On Sat, Jul 08, 2006 at 10:59:49AM +0100, Angus Comber wrote:
I had an Asterisk installation working fine for CallerID on BT analog lines
using a Digium analog 4 port card. However, user switched to TalkTalk
without telling me and CallerID no longer works. However,
Salve Tzafrir!
On Sat, 08 Jul 2006, Tzafrir Cohen wrote:
Right. Don't use safe_asterisk . asterisk daemonises just fine on its
own. asterisk -r gives you a nice console. If you want a complete trace,
you have the logs at your disposal: take a visit at logger.conf.
AFAIK will safe_asterik
Dear all,
I'm desperately trying to get Asterisk working with a FRITZ PCI card on
Debian with kernel 2.6.15.
I'm wondering if anybody has such a working installation.
Thank you for your help, Guy.
Guy Corbaz
ch. du Châtaignier 2
1052 Le Mont
I've been playing with realtime voicemail, and have got everything going
using ODBC. However, I have noticed that the MWI does not come on when
there are messages for me.
If I move back to the static voicemail.conf, it does light up.
Is there any reason for this not to work ?
Julian
Salve Tzafrir, *!
On Sat, 08 Jul 2006, Tzafrir Cohen wrote:
Sytem($( sleep 4 cp 1.call /var/spool/asterisk/outgoing) )
However you should not copy to the outgoing queue. You should mv a file
there.
Right, I remember that a good friend of mine talked about the
disadvantages of the
In my interpretation of the oft confusing Cisco
licensing structure for phones, the license was originally created to function
much like a COA with a piece of Microsoft software. When adding a client
phone to a CallManager or CallManager Express network, the user is required to
have a
Cory Andrews wrote:
In my interpretation of the oft confusing Cisco licensing structure for
phones, the license was originally created to function much like a COA
with a piece of Microsoft software. When adding a client phone to a
CallManager or CallManager Express network, the user is
Its strange, if I reboot my Asterisk you get no callerid. But then if you
do a reload of the config then callerid comes back. any ideas why this
could happen?
Angus
- Original Message -
From: Steve Kennedy [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Saturday, July 08,
On 7/7/06, Robert Michel [EMAIL PROTECTED] wrote:
Salve *!
Semicolon inside extensions.conf marks the start of the comment,
so no chance to use it to build a list of bash comments -
alternative: and || see:
man bash | grep -A 33 Lists
When I try to use my (GSM)mobile to initiate a
Robert Michel wrote:
Salve Tzafrir, *!
On Sat, 08 Jul 2006, Tzafrir Cohen wrote:
Sytem($( sleep 4 cp 1.call /var/spool/asterisk/outgoing) )
However you should not copy to the outgoing queue. You should mv a file
there.
Right, I remember that a good friend of mine
On Sat, Jul 08, 2006 at 12:34:42PM +0200, Robert Michel wrote:
Salve Tzafrir, *!
Thank you for your kindly (and good) support. :)
I found a solution for me to get a colored CLI on a
vserver, but to publish it for other *-users,
I feel it must be smarter ;)
On Tue, 04 Jul 2006, Tzafrir
On Tue, Jul 04, 2006 at 05:10:35PM +0200, Robert Michel wrote:
Salve *!
I'm using asterisk for a while and now I want to have a colord CLI.
I have apt-get install asterisk/testing, that is asterisk 1.2.7.1
I use Debian stable/testing on a vserver with any /dev/tty*.
So, of course, I
On Sat, Jul 08, 2006 at 04:10:54PM +0200, Robert Michel wrote:
Salve Tzafrir!
On Sat, 08 Jul 2006, Tzafrir Cohen wrote:
Right. Don't use safe_asterisk . asterisk daemonises just fine on its
own. asterisk -r gives you a nice console. If you want a complete trace,
you have the logs at
Salve *!
Small improvement, I think it would be a good
idea to use $RANDOM for the temporary file
name - you could never known what it is
running at the same time, especialy in case
that someone would use this script not
for asterisk callfiles -
remember my script is just a try ;)
#!/bin/bash
#
Salve Maxim!
On Sat, 08 Jul 2006, Maxim Vexler wrote:
exten = 100,1,System(touch /tmp/file1 \; touch /tmp/file2) works for me.
Here, too ;)
Erghhh, I should played with \ before send noise to this list...
But thank you - your hint should be add to a documentation of
System()
Note that and
On 17:33, Sat 08 Jul 06, Robert Michel wrote:
Ok \; is working fine - I just have modificated my vim syntax
highlighting:
wanna share that file ?
--
Michiel van Baak
[EMAIL PROTECTED]
http://michiel.vanbaak.eu
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD
Why is it
Salve Michiel!
On Sat, 08 Jul 2006, Michiel van Baak wrote:
On 17:33, Sat 08 Jul 06, Robert Michel wrote:
Ok \; is working fine - I just have modificated my vim syntax
highlighting:
wanna share that file ?
See:
From: Robert Michel [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Ciao Marco,
in your init-misdn.conf (or misdn.conf, not sure now...) you can
choose the MSNs for your incoming Ports or Outgoing ports,
msns=3223242,3223243,3223244
The msns must be set in the asterisk misdn.conf, the misdn-init.conf
file contains the settings of the misdn kernel driver.
On 16:44, Sat 08 Jul 06, Florian Overkamp wrote:
Point is, you do not really need a CH1 or CCME license, you are free to
combine the Spare phone with a separate SIP license - the price is
identical. It is NOT OK however to use a Spare phone without any
license, as far as I am aware.
Thanks
Salve Tzafrir!
On Sat, 08 Jul 2006, Tzafrir Cohen wrote:
But you are right, there are options for [EMAIL PROTECTED]
to work around. My skripting skills are not so high
and my try would be better inside the asterisk scripts
for shure - but I'm looking for a solution that is
On Jul 8, 2006, at 8:18 AM, Tzafrir Cohen wrote:
snip
Are colors that important to you? It shouldn't be complicated to get it
working. I'll have a look.
snip
Actually I have noticed that the color console is major cpu eater,
versus the monochrome one...
I use both screen and ssh to
Salve Martin!
On Sat, 08 Jul 2006, Martin Joseph wrote:
On Jul 8, 2006, at 8:18 AM, Tzafrir Cohen wrote:
snip
Are colors that important to you? It shouldn't be complicated to get it
working. I'll have a look.
snip
Actually I have noticed that the color console is major cpu eater,
Salve Martin!
On Sat, 08 Jul 2006, Robert Michel wrote:
Actually I have noticed that the color console is major cpu eater,
versus the monochrome one...
Good point.
Even when no other asterisk -crvvv is conected?
In case yes, would be an idea that the master asterisk
fall back to
Michiel van Baak wrote:
On 16:44, Sat 08 Jul 06, Florian Overkamp wrote:
Point is, you do not really need a CH1 or CCME license, you are free to
combine the Spare phone with a separate SIP license - the price is
identical. It is NOT OK however to use a Spare phone without any
license, as far
On Fri, 2006-07-07 at 08:28 -0500, Cavanna, Richard wrote:
I am thinking of using this machine to run asterisk. Has anyone had any
experience with this machine?
Have one here.
Get enough mem, default deliverey is with 256 MB.
Draw back is however it needs PC2-4200/5300 ECC DDR2
It has only
On Sat, Jul 08, 2006 at 05:20:31PM +0200, Robert Michel wrote:
Salve *!
Small improvement, I think it would be a good
idea to use $RANDOM for the temporary file
name - you could never known what it is
running at the same time, especialy in case
that someone would use this script not
for
The Cisco licenses are non transferrable. If you bought a phone on Ebay
that was advertised as a Licensed phone, and the original owner registered
the serial number of the unit with Cisco, you are supposed to obtain a new
license, as the registered license, which is tied to the serial number,
Hans Witvliet wrote:
On Fri, 2006-07-07 at 08:28 -0500, Cavanna, Richard wrote:
I am thinking of using this machine to run asterisk. Has anyone had any
experience with this machine?
Have one here.
Get enough mem, default deliverey is with 256 MB.
Draw back is however it needs
Is possible to set an default volume on a PBX, so
that all lines have the same volume, like if some one speaks loud, he would get
a low volume.
//Michael
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users
On Sat, Jul 08, 2006 at 01:55:00PM +0100, Thomas Kenyon wrote:
[snip]
I thought that the line would now go through talktalk (It is an LLU
service after all).
FWIW, the same thing happened to me with a line that moved to bulldog.
In the UK BT still own 85% of all copper into premises. Ofcom
On 7/8/06, Doug Lytle [EMAIL PROTECTED] wrote:
Robert Michel wrote:
Salve Tzafrir, *!
On Sat, 08 Jul 2006, Tzafrir Cohen wrote:
Sytem($( sleep 4 cp 1.call /var/spool/asterisk/outgoing) )
However you should not copy to the outgoing queue. You should mv a file
there.
Right, I
I cant make call when using #31#0046011
The call just disapper and noting shows on CLI
Using asterisk 1.2.9.1
I am using cubix IAX2 softphone
And I had the same problem when I used xlite sip
softphone
My voip-provider is Rixtelecom
//Michael Nielsen
Salve Tzafrir!
On Sat, 08 Jul 2006, Tzafrir Cohen wrote:
To simplify error handling:
set -e
tmpfile=`mktemp`
Note that the temp file has to be in the sme filesystem as the asterisk
spool
It would be more performant when it is the same filesystem,
but it should also work when not. AFAIK will
Hi all,
I have two pda's and I want to be able to make calls, but I need a
client for this. The only problem is Windows Mobile 5.0, I can't find
a freeware client for this, the only one is Sjphone. But this one is
still beta for windows mobile and it just doesn't work good.
Does anyone
Hi everyone,
Can someone post an example of how you read in a channel variable from
asterisk through PHP. I tried the ones voip-info.org but none of them
seem to work, or at least I am not doing something write, but I have no
problem setting variables and other functions, just reading
On Sat, Jul 08, 2006 at 09:39:36PM +0200, Robert Michel wrote:
Salve Tzafrir!
On Sat, 08 Jul 2006, Tzafrir Cohen wrote:
To simplify error handling:
set -e
tmpfile=`mktemp`
Note that the temp file has to be in the sme filesystem as the asterisk
spool
It would be more performant
Can someone post an example of how you read in a channel variable from
asterisk through PHP. I tried the ones voip-info.org but none of them
seem to work, or at least I am not doing something write, but I have no
problem setting variables and other functions, just reading variables
into my
I have tried both ways (with PHPAGI and without), and neither works I
went back to a real simple test, and that doesn't even work.
Here is the CLI:
- Executing AGI(SIP/9897943713-9e04, VoiceMail.php) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/VoiceMail.php
-- AGI Script
On 18:41, Sat 08 Jul 06, Kevin Smith wrote:
Any ideas?
On the asterisk CLI type: agi debug
Try again
--
Michiel van Baak
[EMAIL PROTECTED]
http://michiel.vanbaak.eu
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD
Why is it drug addicts and computer afficionados are both
Salve Tzafrir!
On Sat, 08 Jul 2006, Tzafrir Cohen wrote:
sslash=$(echo $1 | sed 's/[^/]//g')
use 'basename' instead?
Good idea, sp=${1%/*} creats trouble
when there is no slash inside $1
You should exit with an error if you did not deliver a call file.
exit 1
Also, to make this a
What are you using (misdn, capi, something else?) and what problems are you
having?
I submitted a patch recently to mISDN which should have fixed a problem on
hangup, if that's the problem you are having then try the latest cvs mqueue
branch of mISDN.
James
-Original Message-
From:
I have tried both ways (with PHPAGI and without), and neither works I
went back to a real simple test, and that doesn't even work.
Here is the CLI:
- Executing AGI(SIP/9897943713-9e04, VoiceMail.php) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/VoiceMail.php
-- AGI
Hey guys, thanks for the suggestions, I finally figured it out.
I need to run the script using the CGI version of php or
#!/usr/bin/php-cgi -q...not really sure why, but it all started working,
AGI classes and all.
Thanks again,
Kevin
Time Bandit wrote:
I have tried both ways (with PHPAGI
Hi -I hope someone out there can help. I've just built a new asterisk server running [EMAIL PROTECTED] 2.7. and I'm having real difficulty setting up my cable modem for the internet connection. I have 1xcable modem and 1xnetgear router and 1xPCI nic card. I simply set the netgear up as
Al Lougher wrote:
Hi -
I hope someone out there can help. I've just built a new asterisk
server running [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] 2.7. and I'm
having real difficulty setting up my cable modem for the internet
connection. I have 1xcable modem and 1xnetgear router and 1xPCI
Hey guys, thanks for the suggestions, I finally figured it out.
I need to run the script using the CGI version of php or
#!/usr/bin/php-cgi -q...not really sure why, but it all started working,
AGI classes and all.
Strange, I run it with standard PHP
#!/usr/bin/php -q
Well, if it works, then
I agree, that's what every example I saw was using. But ya, it's working
now so I'm a happy camper :D
Time Bandit wrote:
Hey guys, thanks for the suggestions, I finally figured it out.
I need to run the script using the CGI version of php or
#!/usr/bin/php-cgi -q...not really sure why, but
Hope someone call help me .I have 2 POTs line coming into Asterisk. We have callerid feature from Verizon on one of the lines.I am not able to track any CallerID coming in, in the log. I am pretty green with asterisk, and it's not clear if I have to activate for CallerID in the dialplan. The
and what TDM card are you using and what does your zapata.conf file look like.
On 7/8/06, Ryder Brook [EMAIL PROTECTED] wrote:
Hope someone call help me .
I have 2 POTs line coming into Asterisk. We have callerid feature from
Verizon on one of the lines.
I am not able to track any CallerID
As of Asterisk 1.0.X a "#" was recognized as a pattern not as a digit, hence in order to use it at the begining of an extension you should use "_" before it. I guess this is still valid in 1.2.X versions.i.e: use _#31#0046011 in your extensions.confAlyed
Return-Path:
like if some one speaks loud, he would get a low volume.I'm sorry, but this goes far beyond Asterisk (at least for the moment) :)Anyway you can still play with rxgain and txgain in zapata.conf, but this will increase/reduce the overall volume gains and can also affect echo perception.Alyed
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