On Wed, 16 Jan 2013, Zohair Raza wrote:
Make sure Asterisk has access to your AGI script, and make it executable
(chmod u+x agi.php). Also make sure it has shebang (!#/usr/bin/php)
Make sure that the user executing the Asterisk process can execute your
script. Ownership (user and group),
***When you say 'doesn't work' do you mean 'doesn't do what I want' or
'does not execute?'*
I mean I do all steps in Mr. Nir presentation documents and not works.
Here is my php code:
#!/usr/bin/php -q
?php
error_reporting(E_ALL);
ob_implicit_flush(false);
set_time_limit(6);
$stdin =
Hi,
I'm usually working with GN Netcom 9120 Flex and have been very satisfied
with it but for Call Center agents needed to wear and work with headset all
day long in potentially noisy offices, I'm wondering if wireless headets
with bluetooth connectivity exist ?
I'm refering to bluetooth as
On Thu, 2012-01-12 at 11:51 +, Ishfaq Malik wrote:
Hi
I'm using 1.8.7.0 with the RealTime architecture.
If a call goes into application Queue and is abandoned by the caller, no
entry is made in the CDR. Entries are made into the queue log.
This cannot be correct behaviour, all
Hello All,
I am having a bit peculiar problem with Asterisk 11 for a
carrier. This carrier shares quite some information in SDP header, which
should not be the problem, however what happen is as follow:
Carrier (INVITE) - *SIP Proxy - Asterisk 11 - Answer()* - right
after
Le 16/01/2013 12:29, Olivier a écrit :
Hi,
Hello
I'm usually working with GN Netcom 9120 Flex and have been very
satisfied with it but for Call Center agents needed to wear and work
with headset all day long in potentially noisy offices, I'm wondering
if wireless headets with bluetooth
On 01/16/2013 07:28 AM, Salman Zafar wrote:
Hello All,
I am having a bit peculiar problem with Asterisk 11 for a
carrier. This carrier shares quite some information in SDP header, which
should not be the problem, however what happen is as follow:
Carrier (INVITE) - *SIP
On 01/16/2013 05:31 AM, Ishfaq Malik wrote:
On Thu, 2012-01-12 at 11:51 +, Ishfaq Malik wrote:
Hi Everyone
This issue has reared it's ugly head again for us. If a call comes into
a queue and the caller abandons the call, the call does not show in the
CDR.
This is also the case for
Hi Christopher,
I'm using Asterisk 10.4.2. Do I need to install updated version to resolve
this issue? Please advise.
--
Date: Tue, 15 Jan 2013 15:45:31 -0600
From: Christopher Harrington ch...@acsdi.com
Subject: Re: [asterisk-users] Getting UDPTL (SIP):
2013/1/16 Administrator TOOTAI ad...@tootai.net
Le 16/01/2013 12:29, Olivier a écrit :
Hi,
Hello
I'm usually working with GN Netcom 9120 Flex and have been very satisfied
with it but for Call Center agents needed to wear and work with headset all
day long in potentially noisy
2013/1/16 Administrator TOOTAI ad...@tootai.net
Le 16/01/2013 12:29, Olivier a écrit :
Hi,
Hello
I'm usually working with GN Netcom 9120 Flex and have been very satisfied
with it but for Call Center agents needed to wear and work with headset all
day long in potentially noisy
On Tuesday 15 January 2013, Ahmed Munir wrote:
Hi,
I configured Asterisk 10 for inbound fax, for couple of weeks I didn't see
any issues until today. The setup I configured for inbound fax is quite
simple i.e. Cisco Voice GW sends the fax calls to Asterisk using T.38
protocol and later
On Wed, 16 Jan 2013, Muhammad wrote:
**When you say 'doesn't work' do you mean 'doesn't do what I want' or
'does not execute?'
I mean I do all steps in Mr. Nir presentation documents and not works.
Your PHP script executes correctly on my dev box, but I would change the
log file path to
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Yves A.
Sent: Tuesday, January 15, 2013 6:07 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] special conference room
Hi list,
I am in need of a
Please study meetme application's options. You will get almost all feature
you ask for in it
On Jan 16, 2013 5:37 AM, Yves A. yves...@gmx.de wrote:
Hi list,
I am in need of a special asterisk conference room with the following
constraints:
- there is one admin / moderator and several normal
I'm trying to decide if I need to open an issue for this or if it's just a
misconfiguration issue of some sort. Here's the situation - yesterday
morning, I downloaded asterisk 1.8.19.1 and installed it on a fresh CentOS
5.8 installation and got a shell of a basic asterisk install setup (minimum
I am also experiencing this issue. Asterisk is in fact running, you can verify
by running asterisk -rvvv (-r connects to an EXISTING asterisk process) or
using ps.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com]
me too.
regards
El 16/01/2013 13:25, Eric Wieling escribió:
I am also experiencing this issue. Asterisk is in fact running, you can verify by
running asterisk -rvvv (-r connects to an EXISTING asterisk process) or using
ps.
-Original Message-
From:
Same issue exists with 11.2
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bakko
Sent: Wednesday, January 16, 2013 1:27 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
On Wed, Jan 16, 2013 at 1:37 PM, Danny Nicholas da...@debsinc.com wrote:
Same issue exists with 11.2
I've created issue 20945 to track this, at least for 1.8.20.0.
https://issues.asterisk.org/jira/browse/ASTERISK-20945
--
Thanks,
--Warren Selby, dCAP
http://www.SelbyTech.com
barat and danny,
thank you for your input...
I am using asterisk 11.2 and i read about meetme. Yes, it has many
switches and options and
can help me a lot... but as you already said... does _almost_ all
features.. unfortunately I
need ALL the constraints fulfilled... therefore i admit I
2013-01-16 22:10, Yves A. skrev:
barat and danny,
thank you for your input...
I am using asterisk 11.2 and i read about meetme. Yes, it has many
switches and options and
can help me a lot... but as you already said... does _almost_ all
features.. unfortunately I
need ALL the
Sounds like a conference with all attendees permanently muted (except the
moderator).
The moderator uses whisper to communicate with individuals.
--Don
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Yves A.
Sent: Wednesday,
Why doesn't the n priority work in a mysql database??
This way I don't have to re-number everything when I insert a new line...
--
Co-op Vacation Rentals
www.coopvr.com
15218 Summit Ave
Suite #300-354
Fontana, CA 92336
Phone/Fax (855) 760-COOP (2667)
--
n priority is a runtime value set by the dialplan. To use it in a
database, you would have to update the database using something like
dialplan show context.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Roy
From what I read, neither confbridge or meetme have the whisper feature
built-in; This doesn't matter because the moderator would have to use
meetmeadmin or the confbridge equivalent to control the other functions.
The moderator would either need two phones or a phone and a web interface.
Let's
ok,
now i have got some very valuable information to start off with. thank
you all.
i´ll be back to report success or further questions...
just one thing, that i think might be a showstopper that i may have not
explained clear enough...:
muting and unmuting a caller should have the effect,
You could set up the caller meetme where the user presses 1 to
Exit the conference
Whisper to the moderator
Rejoin the conference
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Yves A.
Sent: Wednesday, January 16, 2013 4:35 PM
On 17/01/2013, at 4:35 AM, A J Stiles asterisk_l...@earthshod.co.uk wrote:
Unplug 10.3.22.6, and try pinging it. If something answers, then you
indeed
have a clash. Check your DHCP server configuration, and make sure any
manually-assigned addresses are outside its pool of addresses.
If
Thanks Jordan, for having a look at this matter.
Yes, that is what Asterisk 11 is sending. Here are complete sip debugs from
Asterisk attached. Please refer to IP mapping from OP to have a better
understanding.
Is there any way of getting it off from SIP parser on compile time as I am
not using
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