Hello,
Can SRTP be set on a per device basis?
Thanks,
Elliot
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On Mon, Jul 9, 2012 at 7:32 AM, Patrick Lists
asterisk-l...@puzzled.xs4all.nl wrote:
On 04-07-12 06:45, Chandrakant Solanki wrote:
So, is
http://sourceforge.net/**projects/aterisk-amr/files/http://sourceforge.net/projects/aterisk-amr/files/same
patch
also works in 1.8.13.0??
I don't
On 09-07-12 12:00, Chandrakant Solanki wrote:
Hi
@Patrick, are you using which AMR source, will you please provide me
link, I also tried with 1.8.11 but didn't found success.
I am using sourceforge one
http://sourceforge.net/projects/aterisk-amr/files/
I used the one from sourceforge:
On 09-07-12 07:42, Andrew Colin wrote:
Hi Patrick
Can you possibly guide me as to how you got it working.
I am running the same version of Centos and asterisk.
Did you use a specific kernel?
I did not use a specific kernel version. I got the latest versions from
misdn.eu and installed those.
I think so - set UDP and TLS on allow= in sip.conf, then set transport=udp
on your non-secure devices.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Elliot Murdock
Sent: Monday, July 09, 2012 4:27 AM
To:
- Original Message -
From: Danny Nicholas da...@debsinc.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, July 9, 2012 8:17:25 AM
Subject: Re: [asterisk-users] SRTP Encryption Per Device
I think so - set UDP and TLS on
Hi all
I hope that someone of you can solve this. Right now I'm stuck!
I'm using asterisk with some SIP extensions. Basically I want to
establish a call between desktop voip phone (ext 181) and embedded sip
system (ext 182)
All I can see in CLI is:
== Using SIP RTP CoS mark 5
--
Hi,
I am trying to understand how the asterisk queues timeout works. I want
to set a timeout for each phone ring, and a maximum timeout a caller can
wait in a queue, but I didn't get it working the way I want.
My Queue application call looks like this:
Queue(myqueue,t,,,60);
and my
On 7/9/2012 8:24 AM, Sergio Serrano wrote:
Hi all
I hope that someone of you can solve this. Right now I'm stuck!
I'm using asterisk with some SIP extensions. Basically I want to
establish a call between desktop voip phone (ext 181) and embedded sip
system (ext 182)
All I can see in CLI