[Asterisk-Users] SIP - NATIVE BRIDGE ERROR

[Wes Marderness]

I did 'sip show channels' during call. I saw 2 channels, one for each
device, that where on G729A. I thought that when a Native bridge was done *
was releasing the call. I did not think a call would require 2 G729 Channels
because * is just initializing the session between the 2 devices. I have
done the exact same thing on 0.5.0 with the demo license and did not receive
any error or background noise.

thanks for any help,
Wes

-----Original Message-----
From: asterisk-users-admin at lists.digium.com
[mailto:asterisk-users-admin at lists.digium.com]On Behalf Of Brian West
Sent: Friday, February 06, 2004 7:39 PM
To: Asterisk-Users
Subject: Re: [Asterisk-Users] SIP - NATIVE BRIDGE ERROR


Isn't the demo codec 1 channel only?  Then one side is g729 and the other
is what?

do a sip show channels

bkw

On Fri, 6 Feb 2004, Wes Marderness wrote:

> Hi,
>
> Running Version 0.7.2, I receive the following error when attempting to
> connect two SIP Devices.
>
> WARNING[16399]:rtp.c : 1204 ast_rtp_bridge : codec0 = 524556 is not codec1
=
> 524558, cannot native bridge.
>
> The bridge is made but the quality of the call is bad, a lot of disturbing
> noises in background.
>
> Oddly enough, both devices are using only one codec G729. I also am using
> the demo G729 license for Asterisk. I'm not sure how 2 different codecs
are
> being found.
>
> I saw in ast_rtp_bridge function, that the get_codec function returned
these
> values. Could anyone  tell me where the get_codec function is? Curious as
to
> how this is happening.
>
> Should this problem be added to the bug tracker? The SIP calls are very
bad,
> and I did not experience this problem with 0.5.0 .
>
> Thanks,
> Wes
>
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