Hi
Will the presentations made at Astricom 2014 be made public as recorded videos ?
thanks
Bogdan
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Hello Mathew and everyone,
We are still having reboots on our asterisk servers. The latest 12.6.1
release doesn't fix the issue.
We have the core files of the latest reboots and also debug taken during
the reboot.
We would like to open an issue. What kind of information you need for the
issue?
Hello Experts,
Could anybody pl help resolve my query?
Thanks Regards,
Subbaiah Nachiappan
From: NACHIAPPAN, SUBBAIAH (SUBBAIAH)
Sent: Tuesday, October 28, 2014 6:04 PM
To: 'asterisk-users@lists.digium.com'
Subject: RE: Query on connecting 3G MSC with Asterisk PBX Server via SIP
Interface.
On 29/10/14 12:59 pm, A J Stiles wrote:
Imagine what would have happened to the human race if Ugg the Caveman decided
not to share the secret of making fire with everyone freely, but instead went
around demanding shiny beads with menaces from anyone who just wanted to keep
themselves warm .
On Wed, Oct 29, 2014 at 6:59 AM, Yaron Nachum nachum.ya...@gmail.com wrote:
Hello Mathew and everyone,
We are still having reboots on our asterisk servers. The latest 12.6.1
release doesn't fix the issue.
We have the core files of the latest reboots and also debug taken during the
reboot.
On 10/29/2014 05:50 AM, Bogdan Cristea wrote:
Hi
Will the presentations made at Astricom 2014 be made public as recorded videos ?
thanks
Bogdan
I'll second the request for that, and also ask if the sessions on
Kamailio will be similarly available.
Cheers,
j
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I can confirm that all the videos from AstriCon 2014 will be available at
www.AstriCon.net within about 3 weeks.
On 29 Oct 2014 16:33, Jeff LaCoursiere j...@jeff.net wrote:
On 10/29/2014 05:50 AM, Bogdan Cristea wrote:
Hi
Will the presentations made at Astricom 2014 be made public as
Hello,
I am a newer on asterisk. when I tried to send fax, I can not get success.
After doing a lot of reseach, I decide to ask my question at here.
Instead of using old fax machine, I want my system can send fax via t.38.
Could anyone give me some idea to correct my configuration?
Thanks a
On Tue, Oct 28, 2014 at 11:10 AM, Yaron Nachum nachum.ya...@gmail.com wrote:
Mathew,
When I run 'ps -ef|grep asterisk' the following processes are displayed:
root 6861 1 0 Aug27 ?00:00:00 /bin/sh
/ivr/app/asterisk/sbin/safe_asterisk -U asterisk -G asterisk -C
I am happy to report that
https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Applications+REST
+API has the answer to my dilemma. It seems an app has to subscribe to
channel events before it can receive the events like ChannelVarset...
From:
On 10/29/2014 05:50 AM, Bogdan Cristea wrote:
Hi
Will the presentations made at Astricom 2014 be made public as recorded
videos ?
thanks
Bogdan
I'll second the request for that, and also ask if the sessions on
Kamailio will be similarly available.
Cheers,
j
That would be awesome if
Anybody care to share a script or snippet of what they use to remove
leading and trailing silence from customer recorded files?
I've fiddled with sox to remove the leading; reverse the file; remove the
now leading; and reverse the file again, but I'm not really happy with my
results.
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On Wed, Oct 29, 2014 at 1:21 PM, Murthy Gandikota mgandik...@nts.net wrote:
I am happy to report that
https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Applications+REST+API
has the answer to my dilemma. It seems an app has to subscribe to channel
events before it can receive the events
On Oct 28, 2014, at 5:03 PM, Ben Langfeld b...@langfeld.me wrote:
On 28 October 2014 19:47, Derek Andrew derek.and...@usask.ca wrote:
What is the alternative to the dial plan? Is everyone talking about getting
rid of the statements like:
exten = s,1,
what is the alternative?
Remote
Can we expect a SILK codec for 13 ? Or does the one for 12 work for 13?
sean
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On Wed, Oct 29, 2014 at 5:16 PM, sean darcy seandar...@gmail.com wrote:
Can we expect a SILK codec for 13 ? Or does the one for 12 work for 13?
codec_silk for Asterisk 12 will most likely not work in Asterisk 13. A
number of performance improvements in the media handling in Asterisk
required
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