[asterisk-users] One way audio internal

[Andrew Colin]

I currently am running on a

Intel(R) Xeon(R) CPU E5-2670 v2 @ 2.50GHz



Codec im using is



codec_g729-ast18-icc-glibc-x86_64-core2.so



Kind Regards

Andrew Colin

Converged Data (Pty) Ltd.

Licensed Telecoms Operator : (0258/IECS/JAN/09) (0258/IECNS/JAN/09)





Direct: +27 (0)10 591 4607

Mobile: +27 (0)82 310 3007
Switchboard: +27 (0)10 591 4600
Email: andrew at convergedgroup.net

Web: http://www.convergedgroup.net
75 Witkoppen Road, Northriding, Johannesburg, 2169
P O Box 7246, Weltevredenpark, 1715
This communication is confidential and intended solely for the addressee(s). 
Any unauthorized review, use, disclosure or distribution is prohibited. If 
you believe this message has been sent to you in error, please notify the 
sender by replying to this transmission and delete the message without 
disclosing it. Thank you.E-mail including attachments is susceptible to data 
corruption, interception, unauthorized amendment, tampering and viruses, and 
we only send and receive emails on the basis that we are not liable for any 
such corruption, interception, amendment, tampering or viruses or any 
consequences thereof.



From: Mitul Limbani [mailto:mitul at enterux.in]
Sent: Friday, November 21, 2014 1:04 PM
To: Andrew Colin
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] One way audio internal



Then something to do with your codec selection priority.

On 21-Nov-2014 4:26 PM, "Andrew Colin" <andrew at convergedgroup.net> wrote:

I am using the free g729







Kind Regards

Andrew Colin

Converged Data (Pty) Ltd.

Licensed Telecoms Operator : (0258/IECS/JAN/09) (0258/IECNS/JAN/09)





Direct: +27 (0)10 591 4607 <tel:%2B27%20%280%2910%20591%204607>

Mobile: +27 (0)82 310 3007 <tel:%2B27%20%280%2982%20310%203007>
Switchboard: +27 (0)10 591 4600 <tel:%2B27%20%280%2910%20591%204600>
Email: andrew at convergedgroup.net

Web: http://www.convergedgroup.net
75 Witkoppen Road, Northriding, Johannesburg, 2169
P O Box 7246, Weltevredenpark, 1715
This communication is confidential and intended solely for the addressee(s). 
Any unauthorized review, use, disclosure or distribution is prohibited. If 
you believe this message has been sent to you in error, please notify the 
sender by replying to this transmission and delete the message without 
disclosing it. Thank you.E-mail including attachments is susceptible to data 
corruption, interception, unauthorized amendment, tampering and viruses, and 
we only send and receive emails on the basis that we are not liable for any 
such corruption, interception, amendment, tampering or viruses or any 
consequences thereof.



From: Mitul Limbani [mailto:mitul at enterux.in]
Sent: Friday, November 21, 2014 12:51 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: Andrew Colin
Subject: Re: [asterisk-users] One way audio internal



You probably do not have enough g729 channels license.

On 21-Nov-2014 4:17 PM, "A J Stiles" <asterisk_list at earthshod.co.uk> wrote:

On Friday 21 Nov 2014, Andrew Colin wrote:
> Hi All
>
> We have a strange issue with our hosted asterisk server running on Debian
> Internal calls btween extensions using g729 give one way audio
> As soon as we change the codec to ALAW the issues goes away.
>
> Any ideas how to fix this?
>
> Outbound calls via a trunk work fine with g729

Unless you have serious bandwidth issues, just forget about g.729 and change
to a-law throughout.  A-law is what the PSTN  (in civilised countries)  uses
anyway, so you won't need to transcode  (which chews up processor resources
and risks compromising quality)  for calls to and from the outside world.

If you really need to use g.729 and are outside the USA  (therefore, beyond
the reach of software patents),  there is a free version that you can use --
and this one, better than Digium's offering, comes with the Source Code so 
you
can be sure it isn't doing anything nasty behind the scenes.

But to be honest, you probably are better off just sticking with a-law.

--
AJS

Note:  Originating address only accepts e-mail from list!  If replying off-
list, change address to asterisk1list at earthshod dot co dot uk .

--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
               http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20141121/e53c9379/attachment-0001.html>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: image002.jpg
Type: image/jpeg
Size: 10958 bytes
Desc: not available
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20141121/e53c9379/attachment-0002.jpg>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: image003.jpg
Type: image/jpeg
Size: 10956 bytes
Desc: not available
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20141121/e53c9379/attachment-0003.jpg>