On Feb 27, 2014 10:02 PM, "Paul Belanger"
wrote:
>
> >
> No such thing as 'free open source g729 license', if you actually read
the site:
>
There is regarding the copyright on the code. The fact it is also patent
encumbered is a different issue.
> DISCLAIMER: You might have to pay royalty fees t
On Thu, Feb 27, 2014 at 5:24 PM, Jayson Devor wrote:
> Hello Everyone,
>
> We are looking to transition our 23 channels from testing/lab into
> production. During testing we used the free open source g729 license
> using the instructions found here:
>
> http://blog.manhag.org/2010/05/installing-th
Hello Everyone,
We are looking to transition our 23 channels from testing/lab into
production. During testing we used the free open source g729 license
using the instructions found here:
http://blog.manhag.org/2010/05/installing-the-free-g729-codec-for-asterisk/
A little more about our setup. Al
You may want to check out the 3rd party Asterisk module app_konference.
You can find it at http://sourceforge.net/projects/appkonference. I have
customers using it for the last year or so with very few problems. One
customer is routinely running conferences with 80 - 100 users on a Pentium 4
wi
On 13-02-14 17:33, Steven Wheeler wrote:
*From:*asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonas
Kellens
*Sent:* Thursday, February 13, 2014 7:12 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [aster
On 14-02-27 06:26 AM, JULIAN RUSSELL wrote:
> Hi
>
> I am trying out Asterisk 12 between two servers in order to test the
> 100rel (PRACK) capability. When I set the called server to send Ringing
> (180) and wait before answering the call it is not sending 100rel
> Require in the Ringing message (
Is there a way of temporarily suspending participants in a conference?
Say I have 5 users, A,B,C,D,E and I wish A, B and C to have a discussion in the
confbridge session that D and E can't hear, is there a way to suspend D and E
for a while (whilst they are played music or whatever) and later j
Hi
I am trying out Asterisk 12 between two servers in order to test the 100rel
(PRACK) capability. When I set the called server to send Ringing (180) and wait
before answering the call it is not sending 100rel Require in the Ringing
message (non provisional response) therefore the calling serve
Hi
Re raising this issue as it's still affecting me.
Where is the asterisk server getting port 0 from? We use ARA and port 0 is
neither in the full contact not in the port field of the sip table. Nor is
port 0 in the realtime cache for any peer registering from the IP address
generating the error
Greetings to all.
I am not sure of this is a "user" question or a "business" so apologies
in advance if it should be asked in the business list.
A client of mine has a UK branch that is served by a provider that uses
the Broadsoft solution.
I want to create a sip trunk from a remote asterisk pbx t
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