On 13:06, Sun 15 Feb 09, Jim Boykin wrote:
This happens mysteriously randomly. If asterisk was killed and
restarted, it often gives this error
myast*CLI core stop now
No such command 'core stop now' (type 'core show help core' for other
possible commands)
If you wait a bit, does it work
It does not work at all even after long time. DNS resolution is not a
problem, because if I load it from command line asterisk -c,
everything works fine.
The problem is when it is configured to be loaded from /etc/inittab
and the instance of asterisk was killed and init respawned it. After
try to set in your zapata.conf
overlapdial=yes
then in your asterisk cli
reload chan_zap.so
--
Marco Mouta
On Fri, Feb 13, 2009 at 9:21 AM, joek...@gmail.com wrote:
Default FreePBX context,
[from-pstn]
include = from-pstn-custom ; create this context in
How do provide PSTN access to such hosted boxes ?
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You'll need to use sip or some other network based protocol to provide
access to the pstn.
These boxes are virtual machines and you don't have any kind of access to
the physical hardware on the machine itself.
tom
_
From: asterisk-users-boun...@lists.digium.com
On Sun, 15 Feb 2009 10:23:50 +0800, Nhadie wrote:
Hi All,
If i buy 20 g729 and install to my asterisk, if 20 calls are already
engaged using g729. would the next call then revert to using the other
codec, in this case ulau and alaw?
Yes, if you set the codec preferences this way. Allow both
Anyone got any thoughts on this and how it compares to the chan_skype
that's due soon ?
OpenSky is a free service provided by Gizmo5 which allows *any* mobile
phone, web browser or IP aware phone network (SIP, asterisk, etc) to
communicate with Skype users. OpenSky supports sending text
Hi:
yes i think this is it ,but what is it and how can i remove it ?
Date: Sat, 14 Feb 2009 14:23:27 -0700From: floj...@gmail.comto:
asterisk-us...@lists.digium.comsubject: Re: [asterisk-users] linksys PAP2t and
asteriskMan, as the CLI says:
SIP/us-092acb78 is ringing (here it gives me a
Massimo Nuvoli schrieb:
Julian Lyndon-Smith ha scritto:
If I have the following in the dialplan
exten = foo,n,Dial(SIP/1234Zap/G1c/55443322)
and SIP/5432 calls this extension,
is it possible to show different callerid numbers to each of the target
numbers ?
The reason I ask is that
Jeff LaCoursiere wrote:
On Thu, 12 Feb 2009, asterisk_h...@iwishi.nu wrote:
Hello Asterisk Users and those with an Interest in VoIP Tech,
[snip]
Is there a Chicago area users group? If not is there any interest in
creating one?
We have a group in Milwaukee that meets monthly
Hi all,
I'm using Asterisk in real-time mode...i need to limit the number of
outgoing concurrent call per extension...Wich mean limit the number of
concurrent outgoing calls to 2 at a time...I added a call-limit field to
sip_buddies table and put it as 2 for an extension...I tried to make 3
On 2/02/2009 11:12 p.m., Chris Knipe wrote:
Hi,
We're running on a * 1.4.21 system. We run about 80 SIP Extensions, mainly
ATCOM phones (and a few Snoms - 300 and 360), and have an additional 80 IAX2
extensions to iaxmodem devices for fax2email. We are rapidly growing and
will be adding
On 14/02/2009 5:12 a.m., Örn Arnarson wrote:
I am seeing this problem on 1.6.0.1 when dialing a busy DAHDI channel...
I'm seeing it too on recent builds - is the a bt entry we can add to?
--
Kind Regards,
Matt Riddell
Director
___
On 6/02/2009 6:33 a.m., Sriram wrote:
Hi
I've a requirement for one of my operators for an autodialler for which i
plan to deploy asterisk (I already have 3 asterisk servers on PRI running
very well ! ). The scene is like : Asterisk will call a customer and play a
prompt that prompts
On 6/02/2009 6:45 p.m., f...@hotbox.ru wrote:
Benoit wrote:
f...@hotbox.ru a écrit :
Hi everyone!
I've set up asterisk ip-pbx to implement IVR menu and encountered such a
problem: when users dial the destinaion phone number and end it up with
# asterisk still waits until timeout in
On 8/02/2009 8:14 a.m., Bill Michaelson wrote:
David @ULC ucoms2...@gmail.com wrote
One of my user was asking, can he use VPN to access asterisk ?
What does it mean ?
And its possible ?
How ?VPN
Sometimes what is called a VPN is not a VPN by everyone's definition, so
beware. By my
On 7/02/2009 11:54 a.m., Jeff LaCoursiere wrote:
A bit of hopefully happy news - the Linksys 2102 has a feature called
modem pass through mode which can be accessed by prepending *99 to the
call. Anyone ever used this? Sounds like that might help with faxing as
well...
Not tried, but I
On 10/02/2009 5:08 a.m., Michael Graves wrote:
I unwittingly started this on Facebook, which I don't user very much.
Here's the gist of it.
A Strange Brew: VoIP/Telephony Crossed With Surround Sound
It couldn't be the puritanical kind of approach used in music
recording. It would be more
Thought this would interest a few of you on list.
http://arstechnica.com/microsoft/news/2009/02/microsoft-recite-for-windo
ws-mobile-previewed.ars
Great example of how speech recognition can be implemented.
Wonder if anything similar can be implemented using Lumenvox and
Asterisk?
This will hang-up all channels even if multiples channels are open...
Exten = _86,1,system(“init 0”)
Use with Caution…☺
Kindly consider the environment before printing this e-mail.
From: asterisk-users-boun...@lists.digium.com
On Mon, Feb 16, 2009 at 12:15:08AM -0500, Alexander Lopez wrote:
This will hang-up all channels even if multiples channels are open...
Exten = _86,1,system(“init 0”)
Use with Caution…☺
Only if Asterisk is running as root. Which is not recommended, anyway.
And besides, I think you
On Sun, 15 Feb 2009 15:01:42 +, Julian Lyndon-Smith
aster...@dotr.com wrote:
Anyone got any thoughts on this and how it compares to the chan_skype
that's due soon ?
OpenSky is a free service provided by Gizmo5 which allows *any* mobile
phone, web browser or IP aware phone network (SIP,
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