[asterisk-dev] Weird transcoding error (SIP, local channels): sip_write: Asked to transmit frame type 64, while native formats is 256 (read/write = 64/256)

[Jan du Toit]

Hi.

I have the same problem, it only occurs when using SIP LOCAL channels.

Even if you join a meetme on a local SIP channel you don't hear the 
voice saying 'you are the only person in this conference' asterisk just 
go mad piping out Asked to transmit frame type 64, while native formats 
is 256 (read/write = 64/256) in the console.

I'm using SVN-branch-r38420. The previous guy is using version 1.2.9.1.
I alos have 1.2.5 version on which the SIP LOCAL channels work fine.

To what version must be upgrade/downgrade.

Thanks.
Regards.