On Tue, 2004-12-28 at 16:12 -0600, Me wrote:
Dorn,
Can you give me some details on this linux md driver you mentioned?
Also, you say not to scrap the SATA drives, is this because you think I can
use them with FC1 or because you think I should try Debian? I really don't
want to venture
I have set up asterisk with 3 or 4 phones at home, trying to create a
development lab of sorts. I subscribe to the my wife is my guinea
pig philosophy. What I have noticed is that on my IAXy I get a lot of
echo, but it is spuratic at best. I have not experienced any echo on
the other phones
please disreguard this email. Sorry for the inconvenience
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Correct me if I am wrong, but G729 is not distributed with asterisk. By
default it is not available without a license.
http://www.voip-info.org/wiki-ITU+G.729
You have to compile and install the free implementation to test.
Otherwise it won't work...
Kinda hard to get asterisk to use a codecs
Simplest explanation:
IAX and SIP are protocols that allow devices to talk VOIP. When you
connect to a proxy device via either protocol, the proxy is said to
terminate those protocols.
Brent Clements wrote:
I think I have idea what IAX and SIP termination means, but can
someone explain this
Sounds like it is time for a different router... There are a few routers
out there with buggy nat engines... they are fine when you are doing
typical nat, but if you are trying to do 1:1 nat... get a good router or
make a BSD box to use as a router. I would highly recommend
http://m0n0.ch/wall
How about using the tftp server that comes with the linux distribution
you are using to run asterisk?
Just a thought...
Sean
On Thu, 2004-12-02 at 10:39 -0500, Luke Catranis wrote:
Solar winds... free tftp
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
I've been very happy with Netopia 3386-ENTs.
make a BSD box to use as a router. I would highly recommend
http://m0n0.ch/wall for a great do it yourself router
This was so funny that I had to share it:
m0n0wall is probably *the first UNIX system that has its boot-time
configuration
Not sure if there are many differences between UK and US in terms of DSL
signaling, so I will leave that up to someone more informed. However we
have been using zyxel ADSL modems. They do a fantastic job, however
they are (most of them) dumb bridges.
Generally speaking DSL modem manufacturer
This basically means taht param(changefolder) is not set... my guess
is that the previous screen is not passing this config variable.
Consequently when you get to
$changefolder =~ s/(\w+)\s+.*$/$1/;
The variable is not set... I will look at the code and give you a better
idea... vmail.cgi is
Looks like the script is building a select box (drop down menu)... That
value is the one that is not being set... should build a list like:
INBOX, Old, Work, Family, Friends, Cust1, Cust2, Cust3,
Cust4, Cust5
This for some reason in your setup is not getting passed to the script
in the
From: Sean Cook [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] RE: No Files Seen via vmail.cgi
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=utf-8
Looks like the script
From: Sean Cook [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] RE: No Files Seen via vmail.cgi
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain
Actually, put this at the end of the file...
foreach $key
You are absolutely right... my bad
Keith,
Add this to your script instead of what I had you put
while(){
print $_, \nbr;
}
foreach $key (keys %ENV){
print qq{$key $ENV{$key}br\n};
}
Christopher L. Wade wrote:
Sean Cook wrote:
Ok you have a problem... query string should
I think that all you have to do is where you define the codecs for the
extention/protocol and asterisk will take care of the rest...
[sip2101] [sip2102]
allow=g711allow=g729
Asterisk will make the
We are considering a replacement of a legacy PBX system (merlin). I am
trying to figure out which phones would be best supported with the
fullest set of features. Any recommendations?
Sean
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Easiest thing (as long as filedescriptors are being closed properly) is
to increase the number of allowed open files:
/proc/sys/fs/file-max
This file defines a system-wide limit on the number of open
files for all processes. (See also setrlimit(2), which can be
used
Asterisk does not do anything in this vein.
Simply
% echo somevalue /proc/sys/fs/file-max
a good starting point for this value would be double your existing
value.
% cat /proc/sys/fs/file-nr
will give you your existing max files. I would also suggest doubling
your inodes as well.
%
I don't need to increase the system's imposed limit on the number of
open files. I'm more concerned to see if anyone has run across a
memory or fd leak in asterisk that sucks them all up.
My apologies. If you are looking for leaking fd's in asterisk, I am
afraid I am not much help.
I know that this phone will not do sip as it has limited memory, however
has anyone been successful getting it to work with mgcp/h323?
I mistakenly bought one of these thinking it was a 500 (description
noted the 500 documentation). Anyway... just hoping I didn't blow $150
and that I can at
Polycom IP300 are very in-expensive and have multiple lines.
On Thu, 2004-12-16 at 13:49 +0100, Satchid wrote:
What
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Something else to remember some of the
earlier versions would not recognize more
that 8 characters in the file OS79XX.txt had
to be modified to P0M30300 and file had to be
moved to P0M30300.bin
Then at about POM3-06-00 you are force to
start with the signed firmware.
-Original Message-
The company I work for is looking at vendors for a PBX, one of the
requirements is VoIP. I have been sitting there listening to people
pitch very proprietary implementations of VoIP where you are locked in
to their hardware, their interface...
I know a little bit about asterisk (set up a couple
On Mon, 2004-12-13 at 21:04 -0700, Damon Estep wrote:
That is the difference between a commercial project and an open source
project; you must do your part to add value. Surely you do not expect
glossy ad slicks...
As I stated in my original post... just looking for a starting point.
By the
Why can't you use the settings button? If
you know the password (or using the default
password) you should be able to unlock the
phone and do a hard reset...
Sean
-Original Message-
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
om] On Behalf Of
Randy MacKay
Sent: Thursday,
Which distribution?
Sean
On Mon, 2004-12-20 at 12:05 -0500, Greg - Cirelle Enterprises wrote:
Apparently asterisk cannot reboot gracefully (unattended)
when using realtime
MySQL RealTime: Failed to connect database server asterisk on 127.0.0.1.
Check debug for more info.
Does anyone know of any obscur reference for detecting an incoming fax.
I currently have AMP running and everything else is working great.
Installed the spandsp patches and software... using the default AMP
extensions.conf, I start sending a fax, I hear it pick up and transfer
to voicemail after
that obscure. :)
http://www.voip-info.org/wiki-Asterisk+fax
Look for Zap fax detection
On Tue, 21 Dec 2004 08:52:11 -0500, Sean Cook [EMAIL PROTECTED] wrote:
Does anyone know of any obscur reference for detecting an incoming fax.
I currently have AMP running and everything else is working
Actually it is the default install, no changes yet... Maybe the dial
group getting answered before fax detection...
Sean
On Tue, 2004-12-21 at 07:34 -0700, Jason Becker wrote:
forum.
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That was the problem... dial groups drop the call into a different
context before fax can be detected.
I knew it was something simple that I was missing ;)
On Tue, 2004-12-21 at 09:41 -0500, Sean Cook wrote:
Actually it is the default install, no changes yet... Maybe the dial
group getting
You are correct... typo on my part... extentions_additional.conf are
braught into the extensions.conf from the include... under globals
(didn't do a cut and paste for the context)
Thanks,
Sean
On Tue, 2004-12-21 at 07:52 -0700, Jason Becker wrote:
Sean Cook wrote:
Actually it is the default
Mepis is debian based, so you should be able to apt-get
install your sources that you need, specifically the kernel sources.
Google apt-get kernel source and see what you
get...
sean
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Adam
FinebergSent:
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I have an amportal howto for debian sarge at
http://www.squishychicken.com/index.php?option=com_contenttask=viewid=13Itemid=2
enjoy...
Dane Reugger wrote:
Sounds like good advice - I will. But would prefer to settle on Debian -
I have a how two
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Damon Estep wrote:
Jean-Michel,
I agree with all of your comments, and would be willing to bet $100 that NO
AMOUNT OF GOOGLING will answer this question definitively.
I would almost be willing to take that bet... find your exact
configuration
-users-
[EMAIL PROTECTED] On Behalf Of Sean Cook
Sent: Thursday, January 26, 2006 6:12 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] * point to point t1 solution?
Damon Estep wrote:
Jean-Michel,
I agree with all of your comments, and would
- the disadvantage being
a constant ~1000 packet per second Ethernet flow requires to keep the
channels up.
Won't work...
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Sean Cook
Sent: Thursday, January 26, 2006 6:36 AM
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Ross,
I was a little frustrated with Damon's initial reaction to the post as
well. However, we have moved past this ... This is actually turning out
to be quite an interesting thread, lets not get side-tract.
Regards,
Sean
Ross C wrote:
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just set up an extension that goes straight into voicemail main from
your autoattendant...
if you know your parties extension, please dial...
exten = 770,1,VoiceMailMain()
OK Computer wrote:
Can someone point me to a link describing how to enable
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exten = s,1,Dial(SIP/,20)
exten = s,2,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Dial(ZAP/g0/,20)
yrving rivas wrote:
I will appreciate your help.
Thanks!
Yrving
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Ok... I am having a serious brain fart this evening. IIRC, the next sip
draft addresses shared lines and I thought I remembered something on the
list about support for it in the near future.
I also thought that chan_sccp supported it with in
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I have a merlin legend that is connected to my asterisk PBX via te110p
- - 100D on the Legend. Life is good, except some of their context
features do not work because the legend system does allow those features
from PRI. (eg: voicemail, call pickup,
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care to share with the rest of the class?
Dovid Bender wrote:
To monitor who is doing what we writing a program that
every user can have on thier windows desktop to see
the status of all phones on the system. It's AIM
style. Has several groups.
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The GPL primarily deals with linking to the libraries of a GPL project.
I am not aware of any changes they made directly to asterisk, php,
mysql etc that would bind them to the GPL. However, if they are
using/requiring mysql, then they may have to
gmail was blacklisted this week. Many of the mail servers that do rbls
are dropping mail from gmail.
Sean
On Wed, 2006-02-08 at 17:43 -0500, C F wrote:
Am I the only one having trouble with this list?
Since the begining of the week I have not been receiving mail from the
list like I used to,
I am assuming you are using amp for everything... what you do is set up
4 zap trunks with trunk identifiers for 1,2,3,4
Then go to outbound routes and create your routing rules for each of the
routes and have them use the appropriate trunk.
Sean
On Wed, 2006-02-08 at 17:33 -0500, Nelson
Check your /etc/modules.d/zaptel and make sure you have:
install ztdummy /sbin/modprobe --ignore-install ztdummy /sbin/ztcfg
Sean
Tzafrir Cohen wrote:
On Wed, Feb 08, 2006 at 07:10:16PM -0600, Miguel wrote:
Hi i followed this instructions for installing ztdummy on a 2.6 kernel
(taken
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hmmm... I have not found this documented anywhere... ;) Can you
explain the rationale a bit?
Sean
David K Parker wrote:
My problem is solved by doing the following. Many thanks to Mark
Spencer for clueing me in...
search for the following
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The only way you would need authenticated SMTP is for relaying. My
suggestion would be to not set up sendmail to use a smart host but have
it act as an internet mail server. It will lookup the mx records and
make the sending determinations based on
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I normally don't write emails like this, but the last couple of days
have been very frustrating. We built a system for a customer with the
following hardware:
Asus Vintage (sis chipset) P4 2.8
512M Ram
LSI SATA Raid card (2x80g drives)
Digium
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On Tuesday 14 February 2006 10:56, Sean Cook wrote:
we had originally purchased a tdm2400 with echo cancellation but
couldn't fit it in the chassis.
Spent 3 - 4 hours tracking down the source of the echo to no
avail, enabled mark2 echo cancel
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If memory serves me correctly this has to do with ABE only supporting
that number of watched extensions. You are correct that this is an
artificial limitation and I think someone from digium actually
commented that this should be improved in the
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You have to link it to the mysql libraries... add the following to the
apps/Makefile
APPS+=app_cbmysql.so
app_cbmysql.o: app_cbmysql.c
$(CC) -pipe -I/usr/include/mysql -L/usr/lib/mysql $(CFLAGS) -c
- -o app_cbmysql.o app_cbmysql.c
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FYI... I am running this on 1.2.4 and trunk
Sean Cook wrote:
You have to link it to the mysql libraries... add the following to
the apps/Makefile
APPS+=app_cbmysql.so
app_cbmysql.o: app_cbmysql.c $(CC) -pipe -I/usr/include/mysql
-L/usr/lib
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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
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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
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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
.. 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:
undefined symbol: mysql_store_result Feb 16 13:08:17
WARNING[21558]: loader.c:554 load_modules: Loading module
app_cbmysql.so failed!
On 2/16/06, *Sean
Same setup with two TDM400 (8FXO) running for over a year.
On Tue, 2006-02-21 at 01:37 +0100, Thomas Artner wrote:
Am Tuesday 21 February 2006 00:24 schrieb Marc Archer:
Hi All,
Can someone give me a definite answer as to wether or not you can
reliably run multiple TDM400P's in the
I believe that Centrex is ISDN correct?
Sean
On Tue, 2006-02-21 at 04:55 -0800, Dovid Bender wrote:
I do not know a lot about centrex but I know that most
PBX's support POTS lines (usually for faxing). You can
have them switch over the lines that they send you to
pots and then you can plug
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I have actually modified AMP to store the mac address and auto build the
phone.cfg and 0004XXX.cfg files for ftp. I use the default
username and password for the phones, so litterally all you do is plug
them in...
I will put together a
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Is it possible to pickup a call that is on hold on another extension?
Does anyone have any magic they can share on this topic?
I am struggling to teach call parking at a local shop where we installed
*. It would simplify my life so much if they
Just to through another hat in the ring... I use madplay for mp3s...
[default]
mode=custom
directory=/var/lib/asterisk/mohmp3
application=/usr/bin/madplay -Q -o raw:- --mono -R 8000 -a -12
On Thu, 2006-02-23 at 15:23 -0600, Aaron Daniel wrote:
I'd suggest using the format_mp3 program that's
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what about ARI, it gives web based access to the voicemail and is pretty
good at it... the default vmail.cgi is probably not the best as it has a
gaping security hole that allows anyone to listen to anyone elses
messages :)
Sean
Martin Joseph wrote:
But even the FXO - voip bridging is lacking... you basically dial in
and it answers and provides dial tone for you to dial out your VoIP
service.
It doesn't provide incoming pots termination except to the FXS port.
Sean
On Wed, 2006-03-01 at 01:46 -0800, [EMAIL PROTECTED] wrote:
On Tue, 28
First things first... use the latest version... (that I know of)
http://www.fitawi.com/Asterisk/
second... which part are you having problems with? The web piece? or
the app_cbmysql?
For the app_cbmysql, I have found that the easiest way to work with it
is to incorperate it into
I haven't tried sip yet... been finishing voicemail, but the principal
is the same.
res_mysql.conf
[general]
dbhost = localhost
dbname = asterisk
dbuser = someuser
dbpass = somepass
dbport = 3306
dbsock = /var/run/mysqld/mysqld.sock
extconfig.conf
voicemail = mysql,asterisk,voicemail
; i would
In theory I would say I agree how ever in practice... I have a PBX
(Merlin Legend) that I am connected to via PRI (10 foot pre-fab'ed
cable) and I get intermittent echo on the voip side. There is nothing
in between * and the PBX...
sean
On Fri, 2006-03-03 at 13:42 -0600, Michael Sampson wrote:
To add to the other post... aah or amp actually has a DB that contains
call waiting information. It may have the default setup such that call
waiting is disabled. You should be able to dial *70 and enable it.
Sean
On Tue, 2006-03-07 at 11:33 -0700, Rolf Brusletto wrote:
All - I've been
I am using the odbc set up with postgres right now and it works fine.
http://www.voip-info.org/wiki/view/Asterisk+RealTime+PostgreSQL
has most of the info to get you running. As for meetme, I took the
app_cbmysql stuff for webmeetme and rewrote it for postgres. I am still
testing it, but it
Yes you do need unixODBC before you compile asterisk. Once you have
installed unixODBC , asterisk will compile and offer you the following
modules:
cdr_odbc.so
res_config_odbc.so
res_odbc.so
res_odbc.conf and cdr_odbc.conf are the related config files...
Sean
On Thu, 2006-03-09 at 11:57
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I actually have this working... on a merlin legend R7
zapata.conf
; turn off caller id otherwise it hangs...
usecallerid=no
usecallingpres=no
callwaitingcallerid=no
; drop into the vm context
relaxdtmf=yes
context=from-vm
group = 4
signalling =
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If memory servers me correctly DigitTimeout and ResponseTimeout are
depricated...
try:
exten = s,13,Set(TIMEOUT(digit)=5)
exten = s,14,Set(TIMEOUT(response)=30)
Sean
Robert P. McKenzie wrote:
Hello all. I'm having a problem debugging an IVR
I have a sangoma 104d that is our main pbx now( legacy system died ). I
have replaced every phone in the building and things are going very well.
We have fax working well and calls are routing properly... All is well...
Except for our support modems... we have support people that dial out with
Ok... I have heard this on Digium's PBX in the past, but can't seem to find
it anymore.
There was an IVR that you could dial into and Allison had recorded one of
the funniest messages I have ever heard... you are not the next caller,
hang up not, spend time with your children... it was
with
Asterisk. Has anyone done this that wouldnt mind sharing some of the merlin
specific setup?
Regards,
Sean Cook
[EMAIL PROTECTED]
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I am looking for a cost effective way to drop analog lines from our
asterisk system to support modems and faxes. More than would typically
be done with TDMxxB cards.
I have looked at going with a T1 interface to Channel Bank, but that
just seems like a very expensive way to solve this problem.
My understanding is that the biggest difference is ssl for the web
interface. https :)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Noah Miller
Sent: Wednesday, May 25, 2005 1:08 PM
To: asterisk-users@lists.digium.com
Subject:
Do you need volunteers? My employer has just given the go ahead to
devote my time to a project like this
Sean
On Thu, 2005-05-26 at 18:34 -0700, Mitchel Constantin wrote:
We've collaborated, and are going to work on an advanced GUI client
with a web interface to compliment it, it will be
Just got it working with eyebeam:
in sip.conf under general:
videosupport=yes
allow=h261
allow=h263
shouldn't need per phone config.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Matt Riddell
Sent: Tuesday, May 31, 2005 9:43 AM
To: Asterisk
I just ran into an interesting problem. I have a Polycom IP500 that up
until today has been rock solid. We have started exploring video
conferencing and all of a sudden IP500 goes weird. If I initiate a call
to any other sip device, it works fine. But on an incoming call I get
no sound.
If
If this is mod_perl... count me in...
Sean
On Thu, 2005-06-02 at 19:04 -0500, Andrew Latham wrote:
between this and the biz list is looks like a few of use are
duplicating our work.
should we start a frame work up for this as an included project?
On 6/2/05, snacktime [EMAIL
I am not sure if this is really possible but I figured I would ask
anyway. I have a customer who wants an asterisk system. Currently they
have a BizFon system.
The feature that he really wants is to be able to pick up any line and
have all the stations show up on his phone. Is this possible in
I have been hearing a lot about the new Gnet SIP phones. Is anyone
using them? How do they perform?
Sean
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FYI, Dell is offering a $299 Poweredge 2.8Ghz, 256M with two 80gig SATA
drives. I just bought one and have two TDM400P's loaded. I will let
you know how it goes...
On Wed, 2005-06-29 at 19:57 -0600, Tomas Florian wrote:
I had a recent bad experience with Compaq with asterisk. For some
Which fedora? Core 4 has some issue with the boot images on some
hardware. Best to try text install rather than graphical. Core 1 and 2
had various issues with several motherboards (asus, via..) and needed a
custom boot disk to install.
Fedora, as all redhat distros are compiled for 386,
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I am looking to trade for a new or used Sangoma Analog A200 card with
echo cancellation. I have finished my testing with the OpenSwitch
card and want to test with the sangoma. Anyone out there looking to
do the same?
Sean
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The channels are VPB/X
On Fri, 2006-03-10 at 17:35 -0500, Chuck Fletcher wrote:
Any guidance on how to get my openline4 to get recognized by [EMAIL PROTECTED]
I've got my vpb drivers running, but not sure how to add it as a trunk,
should it be via zap? or is there another way?
Thanks,
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This sounds like a digitmap issue... from your sip.cfg what is your
digitmap set to?
Sean
sdgesa gaeharth wrote:
I am using the latest firmware and bootrom and this is a problem with
all 12 polycom 501s that we have in the office. If I want to
file? In other words, does asterisk tell
the phone what extensions are available and then the polycoms change the
map themselves?
thanks
*/Sean Cook [EMAIL PROTECTED]/* wrote:
This sounds like a digitmap issue... from your sip.cfg what is your
digitmap set to?
Sean
sdgesa
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Check to make sure your minimum message length is very short. You
should be able to view this in the full log.
Sean
Phil Freed wrote:
Asterisk 1.2, Fedora Core 4:
When I leave a voicemail message, it writes the necessary files to
the INBOX:
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Is anyone using the Adit 600 with CMG g729 gateway? We are trying to
come up with a solution for 600+ FXS campus and it appears to have the
highest port density of anything out there...
Any other thoughts?
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I would venture to say that all ITSP suck... some just suck less...
It generally speaking boils down to that fact that internet
connectivity is never full reliable (from a consumer standpoint).
Sure if you want to cough up the money for a T1, you
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Ok... so I spent today getting realtime extensions working, which they
are (for the most part) and apart from forgetting to commit
transactions in postgres and trying to figure out why an extension
won't work, all is well.
The only problem that I am
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Now, here is what I'm not sure of at this moment. For the time
being, is it possible to just pass the PRI through the Asterisk to
the Legend? Will there by any type of dialplans or anything that
need to be created? Will it pass the DID
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Aaron,
I have this working quite well. Are you using FTP? or TFTP...
We are using FTP for about 40 phones and it works like a champ. For
each phone I have...
0004f2030925.cfg
APPLICATION APP_FILE_PATH=sip.ld CONFIG_FILES=phone4710.cfg,
sip.cfg
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I am currently running asterisk 1.0.9 on a system with 2 TDM400P... I
have had fairly good success with it across the board... my only issue
is that I have monkeys who move stuff around and things get unplugged ;)
Jared Davison wrote:
I would
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if you have an zaptel card installed and working... try to do a load
app_meetme.so and see what happens... if it loads successfully... you
should be able to conference also check your modules.conf and make
sure you don't have noload=app_meetme.so
with Asterisk is one of the Oh cool! moments.
On 3/23/06, *Sean Cook* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
wrote:
Now, here is what I'm not sure of at this moment. For the time
being, is it possible to just pass the PRI through the Asterisk
to the Legend? Will there by any type of dialplans
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Do hints work in Realtime asterisk? not finding much on the list
archives or anywhere else for that matter... I have tried using -1
priority as mentioned once or twice but no joy
Thought?
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Is anyone using * to provide voicemail to a definity system? I
understand with the new SMDI functionality in trunk that this will be
easier to provide some of the integration features.
Looking for some hints on the definity setup and anything on the SMDI
side. Anyone with a working
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