Friends,
SIPit is the main interoperability event for all things SIP. It's organized by
the SIP Forum and creates good
feedback to the IETF. Asterisk has been participating in SIPit during many
years and in many variants
- videocaps, Marc Blanchet's IPv6 branch and the standard Digium
added canreinvite=yes
Do add a SIP SET DEBUG IP 192.168.13.114 to the game and see what is
happening.
nothing is sent to that IP address. I then turned on STUN in the Snom
phone and set nat=no for that phone in sip.conf. The IP shown in
subscriptions then changed from the private ip to the
Hello!
I have several problems in the audio one belonging to asterisk at
conferences between ZAP - SIP. I hope that you may help me.
1 Problem
When the audio establishes a call between two canals, some zap and another
sip itself one listens interrupted in one of the senses, exactly in zap sip,
Tilghman,
That's what I finally figured out... my understanding, though, is that it is
preferred to use 'file' over mpg123?
On Sat, Mar 6, 2010 at 1:32 AM, Tilghman Lesher tles...@digium.com wrote:
On Friday 05 March 2010 17:19:06 Matt wrote:
For some reason I have to set the type to 'files'
Have just tried the asterisk setup on an old server that I placed on
our LAN (i.e. server and extensions all on same subnet). BLF worked as
expected, no queued messages. Could NAT be the problem?
John
On 6 March 2010 10:16, John j...@vetsurgeon.org.uk wrote:
added canreinvite=yes
Do add a SIP
I have bragged/connected two asterisk over IAX2 but MOH is not working when I
try to dial directly between them?
MOH is working over SIP but not IAX2 ??? (the two asterisks are connected over
VPN).
Here is the log:
-- Executing [...@extensions:1] Dial(IAX2/home_server-1756,
Hi.
I'm trying to set up a bunch of SIP phones to register in various
domains on the local subnets connected to an Asterisk appliance, yet be
able to support Internet SIP calling.
My config looks like this:
[general]
context=INVALID
allowguest=yes
autodomain=no
...
Hi,
I created a dialplan. But now i want to save the keys that users press. How
can i do?
--
Necati DEMİR
http://blog.demir.web.tr
http://friendfeed.com/ndemir
ndemir ~ demir.web.tr
---
--
_
That solution works fine for the polycom phones because you can set two sip
servers.
However, what can I do for the incoming SIP calls from our voip provider?
We can only set one destination address for the calls.
Eg. u...@sipserver1.ourcompany.com
Many thanks
Dan
-Original Message-
On Sat, 6 Mar 2010, Necati Demir wrote:
I created a dialplan. But now i want to save the keys that users press.
How can i do?
You need to be more specific in what you want to do.
You can use the read() application to save user entry in a variable.
You can assign the ${EXTERN} channel
On 03/06/2010 12:46 PM, Joseph wrote:
I have bragged/connected two asterisk over IAX2
The IAX2 stack has a built-in preference for meekness and modesty.
--
Alex Balashov - Principal
Evariste Systems LLC
Tel: +1 678-954-0670
Direct : +1 678-954-0671
Web: http://www.evaristesys.com/
On Fri, Mar 05, 2010 at 03:25:35PM +, Jeff LaCoursiere wrote:
Considering that was a 1950's era composition, perhaps the copyright has
already expired?
A. it's from 1962.
B. John Cage died on 1992. Which means his works will be in the public
domain as of 2062.
(And then again, as other
On Thu, Mar 04, 2010 at 08:59:10PM -0500, Matt wrote:
I've got a ton of files in doc but not that file.
http://svn.digium.com/svn/asterisk/branches/1.6.2/doc/building_queues.txt
--
Tzafrir Cohen
icq#16849755 jabber:tzafrir.co...@xorcom.com
+972-50-7952406
Hi,
Trying for figure out how to write a custom app for asterisk, without
neccisarrily having to utilize the AGI interface. I want to accomplish this
in the dial plan if possible.
What I am trying to do is to write a application which is like the directory
but runs through all the extensions.
My
On Sat, 6 Mar 2010, Sascha Ferley wrote:
Trying for figure out how to write a custom app for asterisk, without
neccisarrily having to utilize the AGI interface. I want to accomplish
this in the dial plan if possible. What I am trying to do is to write a
application which is like the directory
Hi,
I am looking for an Mail-2-Fax and in a second step Fax-2-Mail-solution
that works via T38 with Asterisk, currently still version 1.4 but it
also should work with 1.6.
For Mail-2-Fax I am thinking that you either have to install a special
printer-driver on your Windows-PC (Mac and Linux
I have a TDM400. Just updated Fedora 12 to kernel 2.6.32. Rebuilt and
installed dahdi-2.2.1.
kernel modules loaded.
lsmod | grep wctdm
wctdm 37233 0
dahdi 194985 1 wctdm
lsmod | grep dahdi
dahdi 194985 1 wctdm
crc_ccitt 1549 2
On Saturday 06 March 2010 09:18:13 pm sean darcy wrote:
I have a TDM400. Just updated Fedora 12 to kernel 2.6.32. Rebuilt and
installed dahdi-2.2.1.
kernel modules loaded.
lsmod | grep wctdm
wctdm 37233 0
dahdi 194985 1 wctdm
lsmod | grep dahdi
On Sat, Mar 06, 2010 at 10:18:13PM -0500, sean darcy wrote:
I have a TDM400. Just updated Fedora 12 to kernel 2.6.32. Rebuilt and
installed dahdi-2.2.1.
kernel modules loaded.
lsmod | grep wctdm
wctdm 37233 0
dahdi 194985 1 wctdm
lsmod | grep dahdi
hi
I have problelm with an asterisk 1.6.2.5 tarbal compiled on CentOS 5.4 and
try to change tones in indications.conf but any setting i have made has no
effect. the tones are by default U.S. and i need to change to hungarian or
greece
--
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