If you review the current asterisk 12 sample pjsip config for extension
6002 (viewable here:
http://svnview.digium.com/svn/asterisk/branches/12/configs/pjsip.conf.sample),
you will find it contains the correct settings for an endpoint behind NAT.
Specifically note that you need rewrite_contact enab
What should the PJSIP configuration be if your external IP address is
dynamic, as is common with most home networks, and probably a lot of
small business networks as well? The external_media_address and
external_signaling_address transport settings are static. It would be
possible to write a scri
Hello,
I am struggling to have a SPA504G to auto answer (for intercom/paging). I
have tried the following SIP headers (not all together), but without luck:
SIPAddHeader(Call-Info:\;answer-after=0);
SIPAddHeader(Call-Info: answer-after=0);
SIPAddHeader(Alert-Info: info=intercom);
SIPAddHeader(Alert
On Wed, Oct 22, 2014 at 1:55 PM, Paul Albrecht wrote:
>
> On Oct 22, 2014, at 11:31 AM, Matthew Jordan wrote:
>
>
> On Wed, Oct 22, 2014 at 11:14 AM, Paul Albrecht
> wrote:
>
>>
>> On Oct 22, 2014, at 10:33 AM, Joshua Colp wrote:
>>
>> > Paul Albrecht wrote:
>> >> Really? Shouldn’t something t
> From: Paul Albrecht
> Here’s a link to the minutes: https://wiki.asterisk.org/wiki/
> display/AST/AstriDevCon+2014
>
> It has you saying: Leif: we're in a transition, moving from dialplan
> model to external control model. Probably need external application
> to be built for us to move compl
On Oct 22, 2014, at 2:13 PM, Scott Griepentrog wrote:
> > is asterisk abandoning the dial plan?
>
> It's clear that there is a desire to have a way of running Asterisk with
> little or no dialplan. While currently there is no way to abandon the
> dialplan as you point out, that could actuall
On Oct 22, 2014, at 2:26 PM, Leif Madsen wrote:
>
>
> On 22 October 2014 14:55, Paul Albrecht wrote:
>
> On Oct 22, 2014, at 11:31 AM, Matthew Jordan wrote:
>> This is an open source project. Communication is done in an open,
>> transparent manner. People should feel like they can bring up
- Original Message -
> Greetings-
> Working with the T.38 gateway functionality that is sparsely
> documented [1], I'm attempting to get the following functional:
> Asterisk calling system -> Asterisk system in T.38 Gateway Mode (box
> in question) -> SIP Provider
> The problem is:
>
> is asterisk abandoning the dial plan?
It's clear that there is a desire to have a way of running Asterisk with
little or no dialplan. While currently there is no way to abandon the
dialplan as you point out, that could actually happen, someday, many years
and versions from now. But even then I
On Oct 22, 2014, at 11:47 AM, BJ Weschke wrote:
> On 10/22/14, 12:14 PM, Paul Albrecht wrote:
>> On Oct 22, 2014, at 10:33 AM, Joshua Colp wrote:
>>
>>> Paul Albrecht wrote:
Really? Shouldn’t something this major affecting the entire Asterisk
community get discussed on the lists? Any
On Oct 22, 2014, at 11:31 AM, Matthew Jordan wrote:
>
> On Wed, Oct 22, 2014 at 11:14 AM, Paul Albrecht wrote:
>
> On Oct 22, 2014, at 10:33 AM, Joshua Colp wrote:
>
> > Paul Albrecht wrote:
> >> Really? Shouldn’t something this major affecting the entire Asterisk
> >> community get discuss
Hi there,
I have an issue , I want to make a video call to a streaming source using
Asterisk .
Someone can help me in this issue please?
Thanks in advance.
--
*Élève Ingénieur INE3 à l'Institut National des Postes et
Télécommunications * *INPT - Rabat - Maroc*
*Responsable de la cellule Aster
On Wed, Oct 22, 2014 at 11:14 AM, Paul Albrecht
wrote:
>
> On Oct 22, 2014, at 10:33 AM, Joshua Colp wrote:
>
> > Paul Albrecht wrote:
> >> Really? Shouldn’t something this major affecting the entire Asterisk
> >> community get discussed on the lists? Any idea what Leif is talking
> >> about whe
On Oct 22, 2014, at 10:33 AM, Joshua Colp wrote:
> Paul Albrecht wrote:
>> Really? Shouldn’t something this major affecting the entire Asterisk
>> community get discussed on the lists? Any idea what Leif is talking
>> about when he says the community is in transition, moving from dial
>> plan mo
Hello,
I've got a bunch outgoing-only SIP trunks "connected" to an asterisk 11 setup.
I've read the following doc [1] stating you can pass username/password
in a dial string.
My goal is to dial from asterisk through one SIP trunk or another
without touching my sip.conf file.
In other words, I'm
Really? Shouldn’t something this major affecting the entire Asterisk community
get discussed on the lists? Any idea what Leif is talking about when he says
the community is in transition, moving from dial plan model to external control?
Here’s a link to the notes posted on the Asterisk wiki:
h
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