Thanks Vardan,
You're right. Running the script under h extension gets me the results I'm
looking for.
On Wed, Dec 22, 2010 at 5:38 PM, Vardan Harutyunyan hvarda...@gmail.comwrote:
Try to use h extension
--
Vardan Harutyunyan,
Senior System Administrator
Enterprise Incubator Foundation
Hi Danny,
For question 1, I think sip show peers is what you want.
Yes, indeed. Thanks.
Though, there's something strange with it, but probably related to
question 2 below.
For question 2,
here are two ways to do it.
I tried both ways but couldn't get it working. In both cases, this is
Hi Don,
Lol,
Thanks, for that :)
Kind Regards
Jess
Jess Hart
__
Langley James IT Recruitment
145-157 St John Street Clayton House
Clerkenwell59 Piccadilly
London
Hi Mark,
Many thanks for this, I was getting worried I had the wrong salary.
I think I will join you all in the US
Kind Regards
Jess
+44 (0)7734109934 UK
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Hello,
For a prospective customer, I need to evaluate the cost of an Alcatel OXE IP
trunking licence price.
The setup would be:
PSTN -E1 Alcatel SIPAsterisk -- SIP Phones
At the moment, only a couple of simultaneous calls are needed between
Alcatel PBX and Asterisk but
On Thu, Dec 23, 2010 at 7:51 AM, Olivier oza_4...@yahoo.fr wrote:
Hello,
For a prospective customer, I need to evaluate the cost of an Alcatel OXE IP
trunking licence price.
The setup would be:
PSTN -E1 Alcatel SIPAsterisk -- SIP Phones
At the moment, only a couple
On Wed, 22 Dec 2010 13:22:47 -0500, Bruce B bruceb...@gmail.com
wrote:
This is a NAT issue like noted before.
Try:
localnet=192.168.0.0/ http://192.168.0.0/24255.255.255.0
instead of:
localnet=192.168.0.0/24
http://192.168.0.0/24Also, make sure you have all your VPN connections as
localnet and
Jess Hart wrote:
Hi Mark,
Many thanks for this, I was getting worried I had the wrong salary.
I think I will join you all in the US
Kind Regards
Jess
+44 (0)7734109934 UK
Come and join the line of 15 Million unemployed!
John Novack
-Original Message-
From:
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{\f0\fswiss\fcharset0 Arial;}
{\f1\fmodern Courier New;}
{\f2\fnil\fcharset2 Symbol;}
{\f3\fmodern\fcharset0 Courier New;}}
{\colortbl\red0\green0\blue0;\red0\green0\blue255;}
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{\*\htmltag19 html
To answer your first question - ${MACRO_EXTEN} is a macro-specific
variable. It's the ${EXTEN} that called the macro, since using ${EXTEN}
inside a Macro would just give you a value of s.
As for your second question, that's pretty easy to do. If every outbound
call needs to be formatted in
On Wed, Dec 22, 2010 at 04:48:33PM +0100, Giorgio Incantalupo wrote:
Hi all,
thanks for answering.
You all are right but I do not really need the codec because my phones
and my Voip lines are all working using g729.
I assume you do not need it as you stated. In this case, configure your
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of ?
??
Sent: Wednesday, December 22, 2010 4:22 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Wise selecting of outgoing channel
Hi.
Hi Gilles,
If someone has a working configuration where...
1) Asterisk and some users are on a private LAN behind a NAT firewall
2) some roadwarriors, behind their own NAT firewall, are allowed to
register with Asterisk, and make/receive calls just like they were in
the office
3) the NAT
Tilghman
This does not make any sense. In the voip-info posting for the h
extension it specifically states that to handle h while in a macro that
the macro needs an h extension. The h extension runs inside the macro
but the CDR data is not being updated correctly. Also the rc-ANSWER entry
in
Vardan
I have not use AEL so it is a bit hard to follow with the formatting the
way it is but it looks like correct.
Please note the h extension only appears to run if a call is connected so
I do not know when the CANCEL would ever be set.
There may be someone else who can speak to this. It
Our Organization has a Panasonic PBX KX-TDA200 with a V-IPGW16 card and
I could not make links with our asterisk server h323 through.
While I set ooh323.conf .. but apparently I adjust certain things in
panasonic PBX, it gives me busy this message:
Executing [904246758277 @ from-internal: 1]
On Mon, Dec 20, 2010 at 5:02 PM, Bryant Zimmerman brya...@zktech.com wrote:
I did an upgrade to the SVN trunk on the 12/9 and when I looked in my mysql
table for CDR's today there are no entries since the update.
I have rebuilt and re-installed and re-started asterisk still no CDR's
flowing to
You can also enter into the CLI in order to see if you can spot any
error regarding cdr_mysql, or 'duplicated value for key...' after hangin
up a call.
There might be a corruption in the 'cdr' table (I've seen this
sometimes). You could try a 'repair table cdr' from the MySQL CLI.
Note:
David
I got the svn trunk again and did a make clean and rebuilt the install and
all started to work again. My guess is that it looks like the mysql client
code was out of sync with the server version.
All is good again.
Bryant
From: David Backeberg
Hi,
I had 2 Asterisk servers connected together in iax with auth=rsa and
proper keys for user and peer in each direction. It worked well till I
upgraded one of them to Asterisk 1.6.13 Since I get No authority found
I thought that problem came from keys as the server with 1.6.13 was
changed
Jose
Thanks for your response. It appears that the issue was that the mysql
client drivers were updated when I installed some mono updates and I had to
recompile asterisk the system was actually writing completely blank entries
for every call. Once asterisk was compiled using the newer mysql
On 7 December 2010 17:47, Steve Davies davies...@gmail.com wrote:
On 7 December 2010 15:00, Steve Davies davies...@gmail.com wrote:
On 7 December 2010 14:17, Lee Archer lee.arc...@thebigword.com wrote:
Hi, try unloading res_timing_dahdi.so then trying again.
Lee
-Original Message-
On Wed, Dec 22, 2010 at 5:14 PM, John Novack
jnov...@stromberg-carlson.org wrote:
Lots of unemployed engineers in the US would be more than happy with 70K, or
even less.
A long period of high unemployment in the US, and world markets is something
many have yet to come to understand.
John
Could anyone recommend some documentation regarding Asterisk 1.8 and the
realtime architecture? Specifically I want to know if it is possible to set
a priority label or to use n as a priority for realtime extensions in
Asterisk 1.8? My understanding is that is not possible with Asterisk 1.4 and
I
2010/12/23 Сикорский Сергей s.sikor...@lanet.ua:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of ?
??
Sent: Wednesday, December 22, 2010 4:22 AM
To: asterisk-users@lists.digium.com
Subject:
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John Novack
Sent: Wednesday, December 22, 2010 5:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: Watkins, Bradley
Subject: Re: [asterisk-users] Vacancy -
On Thursday 23 December 2010 09:16:26 Bryant Zimmerman wrote:
In the voip-info posting
Right here is why you fail. Voip-info is very often wrong. Refer to the
documentation that comes with Asterisk for definitive information. In
this case, the h extension should be in the calling context, not
Dear listers,
I'm facing a puzzling situation with Digium TDM2400 card (12 FXO / 12 FXS).
Once a day or so we detect 1 or 2 zombie FXO channels. These can be either
outbound or inbound calls. I thought this could be related to obsolete DAHDI
or Asterisk versions, so I upgraded to 2.4.0 and
On Thu, 23 Dec 2010 15:54:59 +0100, Jeroen Eeuwes
jeroeneeu...@gmail.com wrote:
In sip.conf I have added for all the remote users the setting
canreinvite=no. The downside to that setting is that Asterisk is
always in the audio path. For my situation that does not really
matter.
Thanks Jeroen.
On Fri, 2010-12-24 at 07:52 +1300, CB wrote:
Could anyone recommend some documentation regarding Asterisk 1.8 and the
realtime architecture? Specifically I want to know if it is possible to set
a priority label or to use n as a priority for realtime extensions in
Asterisk 1.8? My understanding
On Thursday 23 December 2010 12:52:48 CB wrote:
Could anyone recommend some documentation regarding Asterisk 1.8 and the
realtime architecture? Specifically I want to know if it is possible to
set a priority label or to use n as a priority for realtime extensions
in Asterisk 1.8? My
On Thursday 23 December 2010 17:55:40 Carlos Chavez wrote:
On Fri, 2010-12-24 at 07:52 +1300, CB wrote:
Could anyone recommend some documentation regarding Asterisk 1.8 and
the realtime architecture? Specifically I want to know if it is
possible to set a priority label or to use n as a
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