Hello Everyone,
Thank your for your response. There are two critical questions I would
like clarified
kindly:
1) We do not perform any transcoding whatsoever. All recordings, and
voice mail are in G729,
and allow=g729 for all peers and in sip.conf. Is there anything else
we need to perform g729
1) We do not perform any transcoding whatsoever. All recordings, and
voice mail are in G729,
and allow=g729 for all peers and in sip.conf. Is there anything else
we need to perform g729 passthrough. More importantly are we still
liable? Given that most vendors support G729, why do some still
On 2014-02-28 14:04, Tahir Almas wrote:
1) We do not perform any transcoding whatsoever. All recordings, and
voice mail are in G729,
and allow=g729 for all peers and in sip.conf. Is there anything else
we need to perform g729 passthrough. More importantly are we still
liable?
On Friday 28 Feb 2014, Tahir Almas wrote:
As earlier referred following quote from their site
DISCLAIMER: You might have to pay royalty fees to the G.729/723.1
patent holders for using their algorithm
You have to pay royalty fee for using their algorithm and it does not
matter whether
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Thursday, February 27, 2014 7:35 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Realtime Call Queues : call members in
Hello,
am attempting again to resolve an issue with multi-tenancy and the forwarding
to VMs between mailboxes. If in a multi-tenancy environment one uses custom
contexts ie.
[a1-ext1](a1)
mailbox=101@a1
and the associated voicemail.conf entry:
[a1]
101 = 1234,My User
On Thu, Feb 27, 2014 at 10:55 PM, Darryl Moore dar...@moores.ca wrote:
On Feb 27, 2014 10:02 PM, Paul Belanger paul.belan...@polybeacon.com
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
On 2/28/14, Johann Steinwendtner steinwendt...@gmx.net wrote:
On 2014-02-28 14:04, Tahir Almas wrote:
1) We do not perform any transcoding whatsoever. All recordings, and
voice mail are in G729,
and allow=g729 for all peers and in sip.conf. Is there anything else
we need to
Correct, I didn't mention this, since I was assuming OP was talking
about getting it into production. Should have been more clear.
Sorry I should clarify. We were incubated for the testing period
however now will be depolying
for commercial use. That being said, we do feel the need to
Asterisk transcodes at many other points. Inband ringing, audio mixing for
conferences, beep tones. It is naive to think you can passthrough g729 and
never transcode without spending significant amounts of time tracking down each
instance Asterisk would have to transcode.
Over the
Why would you use anything other than Digium's fully licensed and fully
compatable with Asterisk modules?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jayson Devor
Sent: Friday, February 28, 2014 4:04 PM
On 28/2/14 9:04 pm, Jayson Devor wrote:
That being said, will purchasing 23 licenses (one for
each channel that we use), and continue to use the open source g729
sorftware keep us legal?
I know at least half a dozen people who do this so that they can more
effectively balance their licence
For 23 channels I recommend a hardware transcoding card.
We use http://www.sangoma.com/products/d100-30-400-sessions/ I think Digium
also has a transcoding card also.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com]
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