IMHO, having per-Asterisk-version doc is very useful feature of
current wiki for those maintening an heterogeneous set of Asterisk
instances or wanting to study differences between versions.
Regards
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-- Bandwidth and Colocat
Hello,
One very useful feature from previous wiki.asterisk.org is giving top
level/direct access to per version reference doc (see [1 as an
example]).
I hope new site will also bring this feature.
[1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+20+Documentation
Cheers
--
__
Hello,
>From recent experiences with Proneprov and Resource List Subscriptions, I
would like to gather comments on the following enhancements.
These include the following items:
1. Tools for list edition
2. Support of Classical List
3. Phoneprov function to iterate over List items
4. List items l
Hello,
I'm working on request to support SIP trunking with IAD boxes connected to
legacy PBXs.
Those PBXs are using ISDN for dial-in-only remote management.
For successful management sessions, IADs require so-called Clearmode
support (rfc4040).
In my target use-case, a management session involves
2018-08-15 19:00 GMT+02:00 :
>
> Message: 3
> Date: Wed, 15 Aug 2018 11:58:21 -0300
> From: Joshua Colp
> To: asterisk-dev@lists.digium.com
> Subject: Re: [asterisk-dev] ContactStatus AMI Event on PJSIP
> Reregistration
> Message-ID:
> <1534345101.543194.1475072552.34b40...@webmai
Hello,
Reading back this thread which enhances templating behaviour, I would
like to ask if the same kind of improvement could be looked at with
setvar statements in config files.
If I'm not mistaken (I didn't checked with Asterisk 13) , when the
following is applied, variable foo is valued to a
Hello,
If I'm not mistaken, numbered placeholders are not supported by
dialplan's SPRINTF function.
Though this is strictly not blocking, having them would allow concise
expressions such as:
Dial(${SPRINTF(${PJSIPFORMAT},mytrunk,123456789)})
with :
Set(PJSIPFORMAT="PJSIP/%2$s@%1$s")
Set(SIPFORM
> --
>
> Message: 3
> Date: Wed, 10 Sep 2014 11:02:16 -0500
> From: Matthew Jordan
> To: Asterisk Developers Mailing List
> Subject: [asterisk-dev] SIPit 31
> Message-ID:
>
> Content-Type: text/plain; charset="utf-8"
>
> Hey everyone -
>
> Myself, Joshua Colp,
- Original Message -
From: "Brian Candler" <[EMAIL PROTECTED]>
To: "Olivier Krief" <[EMAIL PROTECTED]>
Cc:
Sent: Friday, July 28, 2006 3:05 PM
Subject: Re: [asterisk-dev] Routing data modem calls
On Fri, Jul 28, 2006 at 12:49:14PM +0200, Olivier Krief
Hi,
In many cases, I noticed (though I didn't experienced) data modems could
simply be replaced by serial-to-ethernet converters plus dialup routers.
What do you think of that ?
Regards
___
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a
Hello,
I've tried to send it to asterisk-users, but I have received no
answers so I'm trying here.
I have about 10 Asterisk PBX in production with Bristuff-0.2.0-RC8q
(asterisk 1.0.10) and I want to use Bristuff-0.3 now for the new PBX
I am going to set up.
With Bristuff-0.2.0-RC8q the ISD
I have repeatedly mention this issues, and I keep getting laugh at from
Mark... So I do not think donation to digium will fix the core problem.
Digium want to sell the product like it is rightnow, and have no plan to
do masive change to fix any core problems. They think that if they
start re
Just make yourself one. I got a script that make a nightly export of
asterisk/zaptel... of CVS
I base my personal 'stable' build from that.
Marc O.
Hendrik Visage wrote:
On 4/13/05, Tzafrir Cohen <[EMAIL PROTECTED]> wrote:
On Tue, Apr 12, 2005 at 03:26:59PM +0200, Stefan Gofferje wrote:
H
luded to in my
previous email. Of course there are downsides to everything, if everything
varies from the default the config file will be much larger, but you can
reorder the contexts and add/remove them without worrying what will happen
to the rest of the config
James
On Sat, 10 Jul 20
want to make a cheap schema of a PCI
board that can just do timing great, but I don't think we should spend more
time on something that might never work correctly.
Sorry for my bad English, Im Canadian hehe, but I think, let just move on,
How should we start on a better app_conference ?
Ma
Mark fixed the FAX detection thing recently into the CVS, did you apply the
patch for it ?
- Original Message -
From: "john harragin" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Monday, January 26, 2004 4:29 PM
Subject: [Asterisk-Dev] Fax detection, handling and bridged calls?
> I'
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