I am not sure it i possible to make asterisk listen on two different
ports. I think you could run 2 asterisk instances on separate ports,
but dont know if that is usefult o you.
Or you might be able to do something with Iptables firewall software
to forward from one port to another and just run
Hi,
Does anyone know what happens if you exceed your G729 license
capacity? Lets say you have 10 of 10 licenses being used by a PBX,
then an 11th call comes in set up to use G729.
Does asterisk has the ability to stop offering that codec in the SDP
once the capacity is reached.
Robert
, it might be different for different setups.
On Wed, Sep 24, 2008 at 9:21 PM, Robert McNaught [EMAIL PROTECTED]
wrote:
Hi,
Does anyone know what happens if you exceed your G729 license
capacity? Lets say you have 10 of 10 licenses being used by a PBX,
then an 11th call comes in set up to use
Hi,
Does anyone use the nagios plugin for check_sip against asterisk?
Does anyone have a working example of the command definition and
service definition in the nagios config files?
TIA
Robert
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Seems that this got it working as suggested in the thread - thank you
all for replies.
feature feature.1.name=presence feature.1.enabled=1/
I took out the attendant.uri option as you dont need it. It seems to
be that you can set up a buddy watch for one endpoint using this
option - dont know
Hi,
Can anyone please comment on what the issue may be with this. I am
trying to set up an Polycom IP601 with multiple buddy icons displaying
endpoint status.
I am using a polycom IP601, sip 2.2.2.0084
In the phone1.cfg file I set:
attendant attendant.uri=4158149992 attendant.reg=1/
Using
I contacted Polycom support about this a few weeks ago - their answer
was that is was not possible.
We found that hand-configured phones had to be reset by reset device
setting in the Polycom phone menu then Option 66 in a DHCP server
would override whatever was hand-configured and allow you to
and VoIP, which is ultimately in its infancy and
a risk - we may be the first. Amazon are going to be better at this
stuff than most I feel.
Robert
On Tue, Jul 15, 2008 at 2:14 PM, Eric Chamberlain [EMAIL PROTECTED] wrote:
On Jul 11, 2008, at 12:28 PM, Robert McNaught wrote:
Has anyone
provider and not having the headache of
dealing with the hassle and expenses of hardware, racks, cages etc, it
looks pretty attractive.
Any thoughts?
Robert McNaught
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AstriCon
Hi,
I apologize that this is not directly associated with Asterisk, I have
been trying to solve this, but not having any luck.
Does anyone have a setup with http or https with basic authentication
for provisioning Polycom Phones. We use edgemarc 4500 routers and use
Option 66 to auto-provision
an issue since day one. If FTP not an option for you
Alex
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Robert McNaught
Sent: Friday, June 27, 2008 1:20 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject
of putting configs on Cisco IP Phone 7960, can
they please contact me off list?
I've done the configs via tftp, etc but anything into the speaker/handset
relating to voice doesn't work.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Robert
McNaught
Sent
?
On Thu, May 15, 2008 at 1:35 AM, Philipp Kempgen
[EMAIL PROTECTED] wrote:
Robert McNaught schrieb:
Does anyone know how to apply a style sheet to the polycom automatic
provisioning XML files?
Why should applying a stylesheet be different than for any other
XML files?
Even better, does
:
On Thu, May 15, 2008 at 10:08 PM, Robert McNaught
[EMAIL PROTECTED] wrote:
The way I understood it is that TFTP does not allow you to set a
username and password in a URL
like tftp://username:[EMAIL PROTECTED] is not possible
when setting option 66
Is it not possible to require a username
/sounds/custom/disa_greet3)
exten = valid_login,n,DISA(no-password,from-disa,XXX
614)
exten = valid_login,n(end),Hangup
HTH!
Jason
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Robert
McNaught
Sent: Monday, April 28, 2008 6:31 PM
Try having a look at the settings by running 'lokkit' or
'system-config-security-level-tui' from the command lin - ensure that
the firewall is disabled from there also, and turn off SELinux and see
if that makes any difference.
Robert
On Mon, Apr 28, 2008 at 9:11 AM, Jerry Geis [EMAIL PROTECTED]
the AMI or AGI? It just
seems a little strange to use a database for storing temporary data
such as this?
Thanks in Advance
Robert McNaught
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I would recommend the Plantronics CS70N
On Mon, Apr 7, 2008 at 11:47 AM, Noah Miller [EMAIL PROTECTED] wrote:
Any suggestion for a headset (cord and cordless) for IP601?
Any good (and economical) ones from Polycom or Platronics?
I don't know about cordless, but for corded, I've had great
For this, I would recommend using a smart DHCP device, which supports
the passing of 'option 66' - for example, the edgemarc series of
routers.
With that, you could pass ftp://user1:[EMAIL PROTECTED] via dhcp
in order to provision the phone, and different credentials if you are
concerned about
Hi,
Does anyone know anything about the following?
In a hosted environment where several area DIDs are provisioned on a
single server, how do most carriers establish the origination DID,
number.
Asterisk allows us to modify the CallerID, name, number and DNID
channel variables before dialling
Hi,
Does anyone know anything about the following?
In a hosted environment where several area DIDs are provisioned on a
single server, how do most carriers establish the origination DID,
number.
Asterisk allows us to modify the CallerID, name, number and DNID
channel variables before dialling
Im looking to just test the concept of sending SMS texts from *.
When you say a provider? What kind of provider do you mean?
Robert
On Tue, 2007-12-11 at 17:44 -0600, Greg Oliver wrote:
On Mon, 2007-12-10 at 17:58 -0800, Robert McNaught wrote:
Hi
Does anyone have any recommendations
Hi
Does anyone have any recommendations of an SMS gateway which you can
just sign up for on a pay-as-you-go basis for testing, for use with
Asterisk?
Thanks
Robert McNaught
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not in path
[EMAIL PROTECTED] echo $PATH
/usr/kerberos/bin:/usr/lib/courier-imap/bin:/usr/local/bin:/bin:/usr/bin:/usr/X11R6/bin:/home/admin/bin
Is /sbin in your path?
CP
Robert McNaught wrote:
my problem is that a non-privileged user, eg admin, cannot log in and
connect
, which worked, but is
a hack around the problem and don't believe this is the way
It seems that non-privileged users cannot run commands in sbin, but can
in bin directories
Robert
On Mon, Nov 19, 2007 at 08:51:21AM -0800, Robert McNaught wrote:
Hi,
I have set up asterisk to run as non
to
the /home/asterisk/asterisk-bin/sbin/asterisk file, which worked, but
is a hack around the problem and don't believe this is the way
It seems that non-privileged users cannot run commands in sbin, but
can in bin directories
Robert
On Mon, Nov 19, 2007 at 08:51:21AM -0800, Robert
the need in using a database
to store configs - obviously in a big network with hundreds of users,
the advantages of using a database increases with size.
Robert McNaught
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Thanks mate, this helped a lot
On Nov 24, 2007 4:40 PM, Tilghman Lesher
[EMAIL PROTECTED] wrote:
On Saturday 24 November 2007 14:48:02 Robert McNaught wrote:
Does anyone know if it is possible to use the same database and single
ODBC connection to do both CDR recording with cdr_odbc
Hi,
I am trying to write an application which sends DTMF tones once the
called party answers the call from asterisk.
From the way I understand asterisk dialplans work - the below example
will NOT work as the dial application does not finish and move onto
the next priority once the call is
on func_odbc.
I have both res_odbc.conf and cdr_odbc.conf pointing to the same DSN in
odbc.ini
I am starting to think that this limitation in having a single
connection would stop this being possible in asterisk - does anyone know
otherwise?
Thanks
Robert McNaught
Thanks Tzafrir, I took the stuff out of visudo - it turns out the only
way I could get this working was to create a symbolic link -
/usr/bin/asterisk to point to /home/asterisk .asterisk - using
the link created in /usr/sbin/ would not work for 'asterisk -r'
It seems that all commands in
on that?
As far as I am aware it is only possible to put host=xxx.xxx.com once in
sip.conf
Has anyone got this to work, to have a failover outbound proxy in
asterisk, which automatically fails over?
Thanks :-)
Robert McNaught
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be?
and allowing the user asterisk to connect to the console?
Thanks
Robert McNaught
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Hi,
Trying to figure out what would be the best practice for the following:-
Problem: From the main number IVR, allowing a user to dial a 0 to
reach an endpoint instantly (without pressing pound, or having any
delay), or allowing the customer to type in 1234 then pound to be
transferred
:
Sorry, to answer your question, pattern matching should work just fine
with WaitExten().
Robert McNaught wrote:
The other option is the Background (using the m flag) or WaitExten
application, but cannot get these to work using a pattern match - I am
assuming that this is impossible
appreciated.
Cheers
Robert McNaught
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on: unixodbc(E), ltdl(E) - Cannot get rid of
the XXX?
In the yum repository, there is not a unixodbc.
Anyone know the secret to the dependencies?
Robert McNaught
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asterisk-users
per workstation to avoid having to use the phone as a
switch.
I apologize for this question not being directly related to asterisk,
but since Polycom phones are used a lot with asterisk, it seems a good
place to post ;-)
Robert McNaught
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Alan,
What do you mean by the udev rules?
I previously had asterisk compiled and running as user and group
'asterisk'
zaptel and libpri were compiled and installed using user 'root'
so the zaptel service was root. I had a dependency issue with asterisk
trying to access a file owned by root
response?
Thanks
Robert McNaught
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file exists and zap is starting with errors or
alarms.
No Zaptel interface found.
[Oct 15 10:31:21] WARNING[7036]: chan_zap.c:10026 zap_show_status:
Unable to open /dev/zap/ctl: No such file or directory
localhost*CLI
Cheers
Robert McNaught
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