2012/10/31 Benny Amorsen benny+use...@amorsen.dk
Olivier oza_4...@yahoo.fr writes:
That's the point : to me, casual @pickupmark mechanism don't work with
calls that entered into a queue : the extension rings but you can't pick
the call up with a directed pickup.
(For general pickup,
Hi,
I just want to confirm that my problem is solved now and everything is
working as expected .
I used the patch provided in the following link:
https://reviewboard.asterisk.org/r/2171/
Special thanks to Asterisk development team for great responsibility and
quick reaction.
regards
Hello guys,
i would like to implement authentication for my sip extension with an
openldap server.
Following this guide
http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/ExternalServices_id291590.html
i see a template named [sip] to map the information of sip peers into ldap.
You just need a program(C, PHP, Perl) to query LDAP and update SIP. The
example you list requires realtime, but if you roll your own, you could
update /etc/asterisk/sip.conf and issue an 'asterisk -rx sip reload' to
update when needed.
-Original Message-
From:
2012/10/31 Danny Nicholas da...@debsinc.com:
You just need a program(C, PHP, Perl) to query LDAP and update SIP. The
example you list requires realtime, but if you roll your own, you could
update /etc/asterisk/sip.conf and issue an 'asterisk -rx sip reload' to
update when needed.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Giuseppe Longo
Sent: Wednesday, October 31, 2012 8:47 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk and
With this configuration, the peer doesn't authenticate with ldap, right?
2012/10/31 Danny Nicholas da...@debsinc.com:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Giuseppe Longo
Sent: Wednesday, October
Correct. LDAP can be queried to update the Asterisk configuration, but
Asterisk itself is unaware of LDAP.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Giuseppe Longo
Sent: Wednesday, October 31, 2012 8:58
Based on my knowledge, the general section provides an interface to your
LDAP server and the sipuser section sets up one static user.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Giuseppe Longo
Sent:
I don't want update Asterisk configuration, i want to query LDAP only
for name and secret field.
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I don't understand why in [_general] section of res_ldap.conf i need
to put user and pass when
i want to authenticate my extensions.
2012/10/31 Danny Nicholas da...@debsinc.com:
Based on my knowledge, the general section provides an interface to your
LDAP server and the sipuser section sets up
This allows asterisk to open an LDAP connection. Have you reviewed
res_ldap.conf.sample in the configs folder?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Giuseppe Longo
Sent: Wednesday, October 31, 2012
Yes, but i think that's better to open an LDAP connection with
extensions user and password. Or not?
2012/10/31 Danny Nicholas da...@debsinc.com:
This allows asterisk to open an LDAP connection. Have you reviewed
res_ldap.conf.sample in the configs folder?
-Original Message-
From:
Don't really know. My knowledge scale on this one is 99 percent asterisk 1
percent LDAP.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Giuseppe Longo
Sent: Wednesday, October 31, 2012 9:18 AM
To: Asterisk
I'm running Asterisk 10.7.0 with three sip trunks to my call termination
provider. For the most part everything works great.
However, at apparently random times and usually about 20 mins or so into
the call, the outbound audio stream dies.
The call stays connected and the inbound audio works
Giuseppe wrote:
Yes, but i think that's better to open an LDAP connection with
extensions user and password. Or not?
Better is not the right way to look at it. You questions is
about early or late binding. Early binding requires a dedicated
username and password to connect to LDAP before it
Almost two years ago, a change between how AEL code is built into
Asterisk dialplan between minor versions made clear the need to
provide a sane entry point into AEL subroutines and that's how
AELSub() born.
With Asterisk 11 release, they way [stdexten] at extensions.conf is
invoked changed from
is another way to build Multi Tenant system, have to design like
Company A:
Context Company_A
IVR Company A
Extensions: 101,102,103,104 etc.
Company B:
Context Company_B
IVR Company B
Extensions: 101,102,103,104 etc.
Company C:
Context Company_C
IVR Company
Extensions: 101,102,103,104 etc.
Is it possible to bul multitenant system using some third party opensouce
application My design is like this.
Company A:
Context Company_A
IVR Company A
Extensions: 101,102,103,104 etc.
Company B:
Context Company_B
IVR Company B
Extensions: 101,102,103,104 etc.
Company C:
Context Company_C
IVR
Hi
You will need change the names for your extensions
101-company_a
102-company_a
ETC
On Wed, Oct 31, 2012 at 2:23 PM, Darin Iv adari...@gmail.com wrote:
Is it possible to bul multitenant system using some third party opensouce
application My design is like this.
Company A:
Context
On 31/10/12 6:20 pm, Darin Iv wrote:
is another way to build Multi Tenant system, have to design like
Company A:
Context Company_A
IVR Company A
Extensions: 101,102,103,104 etc.
snip
Is there any particular reason why it needs to be _exactly_ like that?
FWIW, we use companyA-201,
Anyone manage to make one of these work *on* an asterisk server? Have
been researching most of the morning and have only found windows-centric
devices that talk SIP to asterisk (of course). I want one that has a
Linux driver that preferably could be an asterisk channel itself.
WIthout
On 10/31/2012 01:38 PM, Jeff LaCoursiere wrote:
Anyone manage to make one of these work *on* an asterisk server? Have
been researching most of the morning and have only found
windows-centric devices that talk SIP to asterisk (of course). I want
one that has a Linux driver that preferably
On Wed, Oct 31, 2012 at 01:38:54PM -0500, Jeff LaCoursiere wrote:
Anyone manage to make one of these work *on* an asterisk server?
Have been researching most of the morning and have only found
windows-centric devices that talk SIP to asterisk (of course). I
want one that has a Linux driver
On 10/31/2012 01:44 PM, Russ Meyerriecks wrote:
On Wed, Oct 31, 2012 at 01:38:54PM -0500, Jeff LaCoursiere wrote:
Anyone manage to make one of these work *on* an asterisk server?
Have been researching most of the morning and have only found
windows-centric devices that talk SIP to asterisk (of
On 10/31/2012 02:49 PM, Jeff LaCoursiere wrote:
On 10/31/2012 01:44 PM, Russ Meyerriecks wrote:
On Wed, Oct 31, 2012 at 01:38:54PM -0500, Jeff LaCoursiere wrote:
Anyone manage to make one of these work *on* an asterisk server?
Have been researching most of the morning and have only found
On 10/31/2012 02:38 PM, Jeff LaCoursiere wrote:
why not just get a usb headset and use with one of the sip client apps ?
if you're going to the trouble of having a phone to plug in the fxs why
rely on the pc at all ?
use one of the spa type routers and plug the pc into it and the phone
or if
On 10/31/2012 02:00 PM, jon pounder wrote:
On 10/31/2012 02:49 PM, Jeff LaCoursiere wrote:
On 10/31/2012 01:44 PM, Russ Meyerriecks wrote:
On Wed, Oct 31, 2012 at 01:38:54PM -0500, Jeff LaCoursiere wrote:
Anyone manage to make one of these work *on* an asterisk server?
Have been researching
Jeff LaCoursiere j...@sunfone.com writes:
The basic question was has anyone made a USB FXS device work with
asterisk. Now that I have additionally defended my architecture
decisions, can anyone actually answer the question?
The Open USB FXS project is exactly what you want. It seems to be
Stop asking same questions !!!
On Oct 31, 2012 11:54 PM, Darin Iv adari...@gmail.com wrote:
Is it possible to bul multitenant system using some third party opensouce
application My design is like this.
Company A:
Context Company_A
IVR Company A
Extensions: 101,102,103,104 etc.
Company B:
Indeed this is getting ridiculous. This person also called me (!!) for
some free consulting after I had posted the answer a few days ago.
NOTE: We aren't going to engineer your system for you! We as a group will
provide help and some basic code to get you started. If you don't know how
to
Greetings-
I'm running into an issue as follows, in simplified form:
A remote Asterisk box, when registered/peered via SIP to a central server, and
makes a call to that central server, is *sometimes* authenticated and calls go
through properly (via from-internal context), and *sometimes* is
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