Thanks Russ for your response.
Finally found time to do more test on this thread.
I uninstalled DAHDI-complete 2.9.1.1 and installed an older DAHDI version 2.4.1
It worked!
Both READMEs said Digium TE420: PCI-Express quad-port T1/E1/J1 should work.
But it seems that 5th gen TE420 (see below) only
I've seen the following appear in some tests with Asterisk 11.11:
WARNING[3938][C-0003]: chan_sip.c:10535 process_sdp: Rejecting
secure audio stream without encryption details: audio 54908
UDP/TLS/RTP/SAVPF 109 0 8 101
Specifically, it always happens from a Firefox 24 host but it works
Dear all,
I was going through sip.conf file and i am not able to
understand the working and how to test the functionality of below fields.
1.tcpauthlimit
2.tcpauthtimeout
any inputs regarding this will appreciated, thanks in advance
Thanks
SHIVAKUMAR
--
Can someone point me to a good tutorial / explanation of local
channels? I've been using them without really understanding what is
going on, and we all know how dangerous that is!
I've read http://www.voip-info.org/wiki/view/Asterisk+local+channels
but I'm just not quite getting it.
--
On 25-08-14 17:06, Mitch Claborn wrote:
Can someone point me to a good tutorial / explanation of local
channels? I've been using them without really understanding what is
going on, and we all know how dangerous that is!
I've read http://www.voip-info.org/wiki/view/Asterisk+local+channels but
On Mon, 25 Aug 2014, Patrick Laimbock wrote:
https://wiki.asterisk.org/wiki/displa/AST/Introduction+to+Local+Channels
s/displa/display/
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice:
On 8/25/2014 1:33 PM, Patrick Laimbock wrote:
On 25-08-14 17:06, Mitch Claborn wrote:
Can someone point me to a good tutorial / explanation of local
channels? I've been using them without really understanding what is
going on, and we all know how dangerous that is!
I've read
On 8/25/2014 2:36 AM, Brian LaVallee wrote:
Hello,
Here's a fun issue that recently caused me some serious heartache.
Hope this helps others from making the same mistake.
Did you know that the configuration parser supports block-comments.
Like an idiot, I've been highlighting text between
The configuration parser can do a lot of things. Out of curiosity
amongst those reading this - how many of you know about templates?
I use templates and wish the realtime parser would understand them as
well.--
_
--
On 8/25/14, 11:44 AM, Joshua Colp wrote:
On 8/25/2014 2:36 AM, Brian LaVallee wrote:
Hello,
Here's a fun issue that recently caused me some serious heartache.
Hope this helps others from making the same mistake.
Did you know that the configuration parser supports block-comments.
Like an
I love the templates and used them extensively in the beginning of my
asterisk journey. But abandoned them once I went realtime. Moving them
to realtime - WOULD BE AWESOME!
Matt Hoskins | NPG Corp | Systems Architect
816.749.2815 (Internal: ext. 10015)
-Original Message-
From:
On Mon, 25 Aug 2014, Joshua Colp wrote:
how many of you know about templates?
(You may get more replies with a more 'on-target' subject. I lost
interest in 'block comments' but was curious why the thread was still
getting replies.)
Love templates. Use them in extensions.conf, sip.conf,
Holy damn - I didn't know you could use templates in extensions! Mind =
Blown.
Matt Hoskins | NPG Corp | Systems Architect
816.749.2815 (Internal: ext. 10015)
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On
On Mon, Aug 25, 2014 at 11:33 AM, Patrick Laimbock patr...@laimbock.com wrote:
On 25-08-14 17:06, Mitch Claborn wrote:
Can someone point me to a good tutorial / explanation of local
channels? I've been using them without really understanding what is
going on, and we all know how dangerous
On Fri, Aug 22, 2014 at 6:48 AM, chandapure shiva
chandapure.shiv...@gmail.com wrote:
Dear all,
I was going through sip.conf file and i am not able to
understand the working and how to test the functionality of below fields.
1.tcpauthlimit
2.tcpauthtimeout
any inputs
On Fri, Aug 22, 2014 at 6:00 PM, Steve Edwards
asterisk@sedwards.com wrote:
Asterisk 11.11.0 on CentOS 6.5, installed from RPMs. OpenSIPS fronting
Asterisk from a Tekelec T9000.
I'm accumulating stuck channels.
snip
I haven't identified what callers are doing to reproduce the error
Here's my current specific scenario. I have a working call me now
solution on our web site. The customer types in their phone number, it
goes into our normal sales asterisk queue via an AMI action. When the
agent answers the call, he gets a brief announcement then asterisk dials
the
In my dialplan, when I dial 99 it rings SIP/2000
When SIP/2000 answers, it hears 9 every 5 seconds until someone dials 9,
what makes 2 legs been bridged.
My problem is: If I hangup, SIP/2000 continues to hears 9 until someone
dials 9 - it not stops
If SIP/2000 hangup - then the call is ended -
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