On 8/14/2011 6:09 PM, Anton Panetta wrote:
Hello,
Raw stats:
Version:1.8.3.2
OS:Centos 5.6
Special setup: postgre database
I am having a few queue issues with Asterisk specifically relating to
breaking out from queues while on hold.
The intent is that while someone is on hold they
Can we see the config for the queue please?
On Mon, 2011-08-15 at 10:39 +0930, Anton Panetta wrote:
Hello,
Raw stats:
Version:1.8.3.2
OS:Centos 5.6
Special setup: postgre database
I am having a few queue issues with Asterisk specifically relating to
breaking out from queues while on
On Sunday 14 Aug 2011, bilal ghayyad wrote:
Hi All;
The main number is 56725000 and we have DIDs from 5000 to 5999. Now, I need
that if five IP Phones make outside calls, then destination should see only
56725111 so I beleive it is related to the DID 5111 but I do not know what
I have to do
Configure un numero GSM qui est operationel dans ton gateway, les addresse ip
dans chaque tel et numero sip.
Ton gateway au niveau des tel ce l' adresse ip de ton ASTERISK.
Neto Dalima Arcene
00243 816857439
De : A J Stiles asterisk_l...@earthshod.co.uk
À :
On 14 August 2011 08:36, Eric Wieling ewiel...@nyigc.com wrote:
I am having a problem with ${DIALSTATUS} and )=CDR(disposition) disagreeing.
Below is a dialplan snippet and the resulting CLI output. This is running in
an 'h' extension.
Noop(DIALSTATUS=${DIALSTATUS})
I think we have that running, i will look into the configuration tomorrow when
back in the office and let you know what the exact setup is
From: Jesie Paluca jesie.pal...@gmail.commailto:jesie.pal...@gmail.com
Reply-To: Asterisk Discussion
Hi,
If you use freepbx, it can easily be set in the configuration of the
extension using Outbound CID
That way you can define outbound CallerIDs for any given extension, be
sure to put it in brackets like this
1099977420
Where 10 is the area code, but that can be different depending on your
Hi,
Am Montag, den 15.08.2011, 10:18 +1200 schrieb Alec Davis:
If you time the *8 just right so it is being handled during
the end of
the Dial then I got:
[Aug 11 16:26:18] ERROR[18458]: astobj2.c:110 INTERNAL_OBJ:
user_data
is NULL [Aug 11 16:26:18] ERROR[18458]:
Numtodial is the variable that will receive the DTMF input - enternum would
be the prompt played before entry (/var/lib/asterisk/sounds/enternum.wav
(gsm, slin, whatever the codec dictates).
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On
Hey there folks,
I'd sent this to the list last night and got reject email this
morning. Apparently it is always a good idea to have an active
subscription to the list you are trying to post to - just one of those
things. :)
In any case, a new beta version of app_swift is available for Asterisk
is this bug already reported at the issue tracker/jira? Is someone
working on it?
Karsten
https://issues.asterisk.org/jira/browse/ASTERISK-18225
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
On Mon, 2011-08-15 at 15:32 +0100, Paul Hayes wrote:
is this bug already reported at the issue tracker/jira? Is someone
working on it?
Karsten
https://issues.asterisk.org/jira/browse/ASTERISK-18225
That's a different issue to what we have been discussing...
--
Ishfaq Malik
On 15/08/11 15:41, Ishfaq Malik wrote:
On Mon, 2011-08-15 at 15:32 +0100, Paul Hayes wrote:
is this bug already reported at the issue tracker/jira? Is someone
working on it?
Karsten
https://issues.asterisk.org/jira/browse/ASTERISK-18225
That's a different issue to what we have been
Hello List,
i have a small issue regarding to some numbers when i call these numbers
using asterisk i got all times answer_machine ,but when i call these
numbers from my phone or any numbers i get the customer and i can speak
without any issue. The issue just when i call these numbers using
I have 3 Linksys/Cisco 504G phones they keep restarting at what seems
to be random. Sometimes as short as 6 minutes.
FW version is 7.4.3a
I have searched and tried disabling FW check and all related settings.
I also extended all the default 3600 resync checks to a lot longer.
TIA
CF
--
On 8/15/11 10:46 PM, C F wrote:
I have 3 Linksys/Cisco 504G phones they keep restarting at what seems
to be random. Sometimes as short as 6 minutes.
FW version is 7.4.3a
I have searched and tried disabling FW check and all related settings.
I also extended all the default 3600 resync checks to
Try putting a w into your dial command
Now
Exten = s,1,dial(dahdi/1/5551212)
After
Exten = s,1,dial(dahdi/1/w5551212)
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of salaheddine
elharit
Sent: Monday, August 15, 2011 12:20 PM
To:
Hello,
Asterisk 1.4.38
Linux version 2.6.9-89.31.1.EL CentOS
Trying to get variables into a dial plan from AMI. I have tried all
sorts of combinations,entering them after making a connection to ami
through telnet, of the many available examples on voip-info.org such as:
Action: Originate
Could you please share a little sample showing how to get connected to AMI
with php?
Thanks a lot!
Elder
On Mon, Aug 1, 2011 at 3:40 PM, Kevin P. Fleming kpflem...@digium.comwrote:
On 08/01/2011 03:35 PM, Paul Belanger wrote:
On 11-08-01 04:24 PM, Daniel - Asterisk wrote:
?php
function
Hi guys,
Did you get some explanation? I'm suffering the exact issue. It could be I
need some additional dependencies than before?
Regards,
Elder
On Mon, Jul 25, 2011 at 7:59 AM, Soeren Malchow (MCon)
soeren.malc...@mcon.net wrote:
Dear Shaun,
First, thanks for you answer
The installed
On Mon, Aug 15, 2011 at 04:56:04PM -0500, Daniel - Asterisk wrote:
Hi guys,
Did you get some explanation? I'm suffering the exact issue. It could be I
need some additional dependencies than before?
Elder, I'm assuming that you too have no pri show channels? If so are you
running from
Yes, I cannot use 'pri show spans' but I have installed it from sources
(./conifugre, make, make install)
But let me make a new try and I'll let you know.
Thanks for your answer.
Elder
On Mon, Aug 15, 2011 at 5:28 PM, Shaun Ruffell sruff...@digium.com wrote:
On Mon, Aug 15, 2011 at
Hello,
I am troubleshooting a SIP trunk problem. The system is Asterisk 1.8.5.
The problem is can't make any outbound/inbound. It always get Number is
not valid 701.
I tried to figure out the reason the call got dropped and couldn't find
out the solution. I noticed that in the SIP debug
On Mon, Aug 15, 2011 at 2:54 PM, Vahan Yerkanian va...@arminco.com wrote:
On 8/15/11 10:46 PM, C F wrote:
I have 3 Linksys/Cisco 504G phones they keep restarting at what seems
to be random. Sometimes as short as 6 minutes.
FW version is 7.4.3a
I have searched and tried disabling FW check
On 8/15/2011 5:48 PM, john Millican wrote:
Hello,
Asterisk 1.4.38
Linux version 2.6.9-89.31.1.EL CentOS
Trying to get variables into a dial plan from AMI. I have tried all
sorts of combinations,entering them after making a connection to ami
through telnet, of the many available examples on
Just wanted to update this to state that Office 365 does not like Comodo
certificates. I acquired a new certificate from Geotrust which resolved this
problem.
From: o o bj_5...@yahoo.com
To: asterisk-users@lists.digium.com asterisk-users@lists.digium.com
Howdy,
I'm working on a macro that authenticates the calling extension against a
list of allowed extensions but it looks like the Expression I'm attempting
to send of pipe separated extensions is showing up as additional arguments
to my macro.
I expected to have 4 arguments to the below macro,
On 11-08-15 06:28 PM, Shaun Ruffell wrote:
On Mon, Aug 15, 2011 at 04:56:04PM -0500, Daniel - Asterisk wrote:
Hi guys,
Did you get some explanation? I'm suffering the exact issue. It could be I
need some additional dependencies than before?
Elder, I'm assuming that you too have no pri show
Trying to make this work, and Office 365 support is useless, giving me the
following response when I asked them for help troubleshooting a 488 Not
Acceptable Here.
Regarding
your service request about configuring your
PBX system with Office 365, we do not support specific setups for PBX
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