On Friday 09 December 2011 12:24:02 am Jeremy Kister wrote:
On 12/9/2011 12:55 AM, Mike Diehl wrote:
What am I doing wrong?
perhaps try: http://jeremy.kister.net/code/asterisk/mwi.pl
I got this to work on the same LAN, but it will not traverse a NAT. At least
it won't for me.
By making
On 09/12/11 01:10 AM, Mike Diehl wrote:
Hi all,
I'm getting ready to start storing all of my voicemail in a mysql database.
I've already got RT sip and RT voicemailboxes working.
I understand that vm storage only works via odbc. I've read all of the
documentation I could find, including:
On 12/07/2011 10:16 AM, Jerry Geis wrote:
I am using AMI to call a phone and play a wave file. That works fine
to SIP/401.
Now I am trying to redirect that call that is ringing to another
phone (SIP/404).
When I do it the other phone rings but the first phone continues to
ring also.
Then
Hi
I'm already mixing asterisk.conf and asterisk realtime architecture
(Macros go in .conf) but is it possible to have a Macro in .ael, another
in .conf and have them both be callable from the realtime database?
Thanks
Ish
--
Ishfaq Malik
Software Developer
PackNet Ltd
Office: 0161 660 3062
On 09/12/11 08:27 AM, Ishfaq Malik wrote:
Hi
I'm already mixing asterisk.conf and asterisk realtime architecture
(Macros go in .conf) but is it possible to have a Macro in .ael, another
in .conf and have them both be callable from the realtime database?
asterisk.ael just gets compiled into
Hi,
I'm trying to debug an asterisk 1.6.1.18 powered system equiped with
an HA-8 and 2 B400M modules (8 BRI ports).
Config is :
libpri-1.4.12
dahdi-2.5.0
Only 4 BRI ports were connected in PtP mode, to 4 telco lines.
This system operated correctly until it suddenly started to spits 245
lines
2011/12/8, Shaun Ruffell sruff...@digium.com:
On Thu, Dec 08, 2011 at 06:41:46PM +0100, Olivier wrote:
2011/12/8, Shaun Ruffell sruff...@digium.com:
On Thu, Dec 08, 2011 at 09:26:14AM +0100, Olivier wrote:
2. Is it normal to see this IRQ changing from time to time ?
Normally, after
On Fri, Dec 09, 2011 at 05:11:47PM +0100, Olivier wrote:
2011/12/8, Shaun Ruffell sruff...@digium.com:
On Thu, Dec 08, 2011 at 06:41:46PM +0100, Olivier wrote:
2. On a more general plan, is taking down layer 1 on idling spans
something PBXs are negociating with each other (the public
On Thu, Dec 08, 2011 at 04:47:37PM -0600, Asterisk Security Team wrote:
[...]
Description It is possible to enumerate SIP usernames when the general
and user/peer NAT settings differ in whether to respond to
the port a request is sent from or the
On Fri, Dec 09, 2011 at 04:52:53PM +0100, Olivier wrote:
Summing all this:
at 18:49:57 INFO: task b400m-0:13594 blocked for more than 120 seconds.
at 18:55:20 NOTICE[2608] chan_dahdi.c: PRI got event: HDLC Abort (6) on
Primary D-channel of span 4
at 00:26:43 NOTICE[2606] chan_dahdi.c: PRI
On Saturday, December 10th, 2011, the Asterisk community services
listed below will be undergoing maintenance (software upgrades and
updates). The services will be shut down at approximately 9:00 PM CST
(3:00 AM December 11th UTC), and will return no later than 10:00 PM
CST. We apologize in
Yes, DAHDI is a timing source and meetme depends on DAHDI for voice
mixing. You can check details here
http://www.asterisk.org/docs/asterisk/trunk/applications/meetme
Install DAHDI then !!?
On Thu, Dec 8, 2011 at 12:46 PM, Durgesh Mishra
durgesh.mis...@rancoretech.com wrote:
In make
You have to install the dahdi framework, but no actual device is required
(dahdi-dummy is loaded).
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Javaid ITEL
Sent: Friday, December 09, 2011 10:52 AM
To:
On Fri, Dec 09, 2011 at 10:55:07AM -0600, Danny Nicholas wrote:
You have to install the dahdi framework, but no actual device is
required (dahdi-dummy is loaded).
Minor clarification: Since DAHDI-Linux v2.3.0 dahdi_dummy.ko is no
longer loaded. You just need dahdi.ko loaded by itself and it
Il 09/12/2011 17.11, Olivier ha scritto:
2. On a more general plan, is taking down layer 1 on idling spans
something PBXs are negociating with each other (the public switch
trying to take the layer one down, listening to an acknowledge
from the private PBX) or is it more brutal than that ?
The Asterisk Development Team has announced the third release candidate of
Asterisk 10.0.0. This release candidate is available for immediate
download at
http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 10.0.0-rc3 resolves several issues reported by the
community
For the sake of posterity, I'm posting this solution:
When I checked the server, the PnP OS option in the BIOS was set to
No. Changing the option to Yes and rebooting has solved the problem.
On 12/8/2011 10:58 AM, Brent Davidson wrote:
I am still having issues with the error message
Dec
2011/12/9, giovanni.v i...@keybits.org:
Il 09/12/2011 17.11, Olivier ha scritto:
2. On a more general plan, is taking down layer 1 on idling spans
something PBXs are negociating with each other (the public switch
trying to take the layer one down, listening to an acknowledge
from the
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