I have the following set in sip.conf [general] section.
rtptimeout = 60
rtpholdtimeout = 300
I would like to set these to default, or null the general settings for one
upline friend as it is solely a fax peer (T38 over SIP)
How can this be easily done?
Michael
On Sat Apr 25 2009 10:35:21 GMT+1000 (EST) Mike Gurson mgur...@gmail.com
wrote:
I am running AsteriskNOW as a VM using OpenSUSE 11 and Xen.
Download the .iso to the local disk and point the installation source
at that. Also, make sure to choose full virtualization NOT
paravirtualized
On Sat, Apr 25, 2009 at 04:43:59PM +1200, Michael wrote:
c/c'd to the Asterisk list as this is probably relevant to Asterisk as well.
My detailed study of the operation of the 'outgoing' directory reveals that
TXFax() does not delete an expired fax batch file (In the 'outgoing'
directory)
Olivier oza-4...@myamail.com writes:
So when receiving 302 Moved Temporarily, Asterisk (version 1.6.1-rc4) is
issuing a new INVITE and doesn't set any BLINDTRANSFER variable.
Thinking back about that, I would say it should have done so.
Your opinion ?
Would you classify that as an attended
We're getting a new server. I'm considering installing 64bit fedora
rather than the 32bit we use now. Is 64 bit a problem with asterisk? Any
issues we should expect?
sean
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On Sat, 2009-04-25 at 06:03 -0400, sean darcy wrote:
We're getting a new server. I'm considering installing 64bit fedora
rather than the 32bit we use now. Is 64 bit a problem with asterisk? Any
issues we should expect?
FWIW I am using 64Bit Debian all the time - works like a charm.
Conrad
Hello All,
I'm a newbie and just started working on asterisk.I have recently installed ADM
and I want to know about ADM (Asterisk Desktop Manager) like its benchmarks
,issues or bugs, compatibility with which asterisk version e.t.c. and then any
Good web-based CRM recommendations.
thanx in
Production setup, no problems whatsoever with 1.6.0.9.
# cat /etc/redhat-release
CentOS release 5.3 (Final)
# uname -a
Linux *** 2.6.18-92.1.18.el5 #1 SMP Wed Nov 12 09:19:49 EST 2008 x86_64
x86_64 x86_64 GNU/Linux
# cat /proc/cpuinfo
processor : 0
vendor_id : AuthenticAMD
cpu
Suggest you use CentOS rather than Fedora.
CentOS has a longer support life, with the same cost.
JMO
John Novack
sean darcy wrote:
We're getting a new server. I'm considering installing 64bit fedora
rather than the 32bit we use now. Is 64 bit a problem with asterisk? Any
issues we should
Anyone thought about something like outgoing queues?
I mean, having same info that has for inbound queues but for outbound calls,
and grouping members there.
For example, before using dial application put an app outqueue that get all
the statics.
Talked time, member status, last call, completed
Sebastian wrote:
Anyone thought about something like outgoing queues?
Many people have. I know QueueMetrics has methods for this kind of
thing, and I'm fairly sure that Vicidial does as well.
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John Novack wrote:
Suggest you use CentOS rather than Fedora.
CentOS has a longer support life, with the same cost.
JMO
John Novack
sean darcy wrote:
We're getting a new server. I'm considering installing 64bit fedora
rather than the 32bit we use now. Is 64 bit a problem with
There's a boat-load of articles on the web with step-by-step guidance. The
first I became aware of was
http://ronaldlewis.com/asterisk-pbx-on-amazon-ec2-how-to-guide-almost-complete/,
another good one is
http://voxilla.com/2009/2/13/asterisk-amazon-ec2-1178
Google is your friend.
On Sat, Apr 25,
On Wed, Apr 22, 2009 at 09:20:05PM +, Jeff LaCoursiere wrote:
I have been wondering - if you ran your SIP traffic over VPN tunnels, what
would the state think of that? They obviously won't be able to inspect
the data to see what is flowing through the tunnel. Do thye also restrict
On Fri, Apr 10, 2009 at 09:15:50PM -0400, Marc Charbonneau wrote:
Didn't try them myself, but I found those 2
- http://x100p.com/products/FXS.php
- http://www.atcom.cn/En_products_AG188N.html
I bought an Atcom AG188N ATA recently and it supports both SIP and IAX2,
has a built in router,
Dears
My scenario is incoming call to asterisk which asterisk in its term will
dial it through its trunk .
I recognized that Asterisk is sending two invites to My Trunk GW IP as you
can see in the debugging below
The first is the default and the second when asterisk receives a 200 OK
Why
Hi,
This message was also sent to the list on the 23rd. Perhaps your mail
client is suffering from a 'double post' fault. Maybe it is similar to
Asterisk's double invite affliction. In fact, looking back at your
post history this is not uncommon and you appear to post the same
questions
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andrew
Joakimsen
Sent: April-25-09 12:28 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] cheap CHEAP ata
Google
On Sat, 25 Apr 2009, Yahya Mohammad wrote:
On Wed, Apr 22, 2009 at 09:20:05PM +, Jeff LaCoursiere wrote:
I have been wondering - if you ran your SIP traffic over VPN tunnels, what
would the state think of that? They obviously won't be able to inspect
the data to see what is flowing
On Apr 24, 2009, at 3:30 PM, Aryan Ameri wrote:
Has anyone been able to get asterisk 1.6 running under Xen or Amazon
EC2?
If yes, can you share your experience please? Is it usable in a
production
environment? How is the sound quality? Am I likely to suffer from
latency
issues if
Jeff LaCoursiere wrote:
On Sat, 25 Apr 2009, Yahya Mohammad wrote:
On Wed, Apr 22, 2009 at 09:20:05PM +, Jeff LaCoursiere wrote:
I have been wondering - if you ran your SIP traffic over VPN tunnels, what
would the state think of that? They obviously won't be able to inspect
Sending works but on receive it keeps failing - reporting back 'training'
failure.
I am using Asterisk 1.6 with T38.
What should I post to the list to assist diagnoses?
Michael
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On Sun, 26 Apr 2009, Steve Underwood wrote:
They can't see the data due to encryption, but they can observe packets'
traffic patterns, and match them to typical usage by various protocols.
Bitek (
http://www.bitek.com/index.php?option=com_contentview=articleid=74Itemid=60
)
is one
On Sun Apr 26 2009 02:48:13 GMT+1000 (EST) Kai-Uwe Jensen kujen...@gmail.com
wrote:
There's a boat-load of articles on the web with step-by-step guidance.
The first I became aware of was
http://ronaldlewis.com/asterisk-pbx-on-amazon-ec2-how-to-guide-almost-complete/
, another good one is
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