On 24/10/2014 12:49 AM, Tim Nelson wrote:
- Original Message -
On 23/10/2014 10:07 PM, Larry Moore wrote:
On 22/10/2014 11:23 AM, Tim Nelson wrote:
Greetings-
Working with the T.38 gateway functionality that is sparsely
documented
[1], I'm attempting to get the following functio
On Thu, Oct 23, 2014 at 10:45 AM, Paul Albrecht
wrote:
>
> On Oct 23, 2014, at 1:55 AM, Olle E Johansson wrote:
>
> > It is critical that a group of developers ask themself questions along
> > these lines - "what if???"
> >
> > - What if we removed AGi and AMI?
> > - What if we made a pluggable
On Thu, Oct 23, 2014 at 3:32 PM, Dave Fullerton <
dfullertaster...@shorelinecontainer.com> wrote:
> Hello all,
> I'm setting up a couple of test boxes and I'm running into a problem.
> What I need help with is determining whether I'm going something wrong or
> if I need to post a bug report. I h
Hello all,
I'm setting up a couple of test boxes and I'm running into a problem.
What I need help with is determining whether I'm going something wrong
or if I need to post a bug report. I have two asterisk 13.0-beta 3
machines set up with extensions connected to each as such:
3700 > AS
> From: Paul Albrecht
> Seems like now is as good a time as any to raise these issues, in
> fact, sooner is better than later because once developers start down
> a path it’s very difficult to get them change their minds no matter
> how much sense it makes. The fact that developers are even
>
On 10/23/2014 11:26 AM, sean darcy wrote:
Running 11.13.1 on Fedora.
This is a new install, but a copy of a previous - working -install.
module load chan_sip
Unable to load module chan_sip
Command 'module load chan_sip' failed.
SIP channel loading...
[Oct 23 14:46:08] NOTICE[669]: chan_sip.c:31
- Original Message -
>
>
> On 22/10/2014 11:23 AM, Tim Nelson wrote:
> > Greetings-
> >
> > Working with the T.38 gateway functionality that is sparsely
> > documented
> > [1], I'm attempting to get the following functional:
> >
>
> What type of endpoint are you using which is originatin
- Original Message -
>
>
> On 23/10/2014 10:07 PM, Larry Moore wrote:
> >
> >
> > On 22/10/2014 11:23 AM, Tim Nelson wrote:
> >> Greetings-
> >>
> >> Working with the T.38 gateway functionality that is sparsely
> >> documented
> >> [1], I'm attempting to get the following functional:
> >>
- Original Message -
>
>
> On 23/10/2014 3:55 AM, Tim Nelson wrote:
> > - 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 sys
- Original Message -
> On 10/22/2014 03:55 PM, Tim Nelson wrote:
> > - 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 -
On 23/10/2014 10:07 PM, Larry Moore wrote:
On 22/10/2014 11:23 AM, Tim Nelson wrote:
Greetings-
Working with the T.38 gateway functionality that is sparsely documented
[1], I'm attempting to get the following functional:
What type of endpoint are you using which is originating the call a
On Thu, Oct 23, 2014 at 10:07 AM, Jared Terrell
wrote:
> with the below defined in logger.conf on 11.6 cert 6
> I am not getting any log message other than notice and warning in any files
>
> when doing module reload logger - queue log is the only one that says it
> restarts
>
> *CLI> module relo
On Oct 23, 2014, at 1:55 AM, Olle E Johansson wrote:
> It is critical that a group of developers ask themself questions along
> these lines - "what if???"
>
> - What if we removed AGi and AMI?
> - What if we made a pluggable PBX?
> - What if we restarted working on a SIP channel?
> - What if w
On Oct 22, 2014, at 3:39 PM, Matthew Jordan wrote:
>
> 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:
>>
On Oct 22, 2014, at 3:27 PM, Kevin Larsen
wrote:
> > 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. Pro
Running 11.13.1 on Fedora.
This is a new install, but a copy of a previous - working -install.
module load chan_sip
Unable to load module chan_sip
Command 'module load chan_sip' failed.
SIP channel loading...
[Oct 23 14:46:08] NOTICE[669]: chan_sip.c:31438 reload_config: Unable to
load config s
with the below defined in logger.conf on 11.6 cert 6
I am not getting any log message other than notice and warning in any files
when doing module reload logger - queue log is the only one that says it
restarts
*CLI> module reload logger
== Parsing '/etc/asterisk/logger.conf': Found
Asterisk Q
Hi,
I use a simple scheme:
SIP video phone A (h264/Asterisk 1.8.11) <---IAX2 trunk> SIP video
phone B (h264/Asterisk 11.7.0)
When calls from A to B and vice versa drop on pickup.
On B side:
[Oct 24 16:33:49] DEBUG[15590][C-0012] res_rtp_asterisk.c: Setting the
marker bit due to a sourc
On 22/10/2014 11:23 AM, Tim Nelson wrote:
Greetings-
Working with the T.38 gateway functionality that is sparsely documented
[1], I'm attempting to get the following functional:
What type of endpoint are you using which is originating the call and is
it T.38 capable?
Larry.
--
_
On 23/10/2014 3:55 AM, Tim Nelson wrote:
- 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
On 10/22/2014 03:55 PM, Tim Nelson wrote:
- 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)
On 23/10/2014 6:41 PM, Larry Moore wrote:
Listing from my Asterisk:
'8000' => 1. Set(SIP_CODEC=alaw)
2. Dial(MulticastRTP/linksys/224.168.168.168:45678/224.168.168.168:6061)
3. Hangup()
Hmm, my SPA525G doesn't auto-answer a page however my SPA92X do.
Just upgraded the firmware on
On 23/10/2014 4:57 PM, Larry Moore wrote:
On 23/10/2014 5:43 AM, Leandro Dardini wrote:
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);
SIPAdd
On 23/10/2014 5:43 AM, Leandro Dardini wrote:
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);
SIPAddHead
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