[asterisk-ss7] Version 0.8.3 of chan_ss7 for asterisk released

[Anton]

Hi Kai,

It's really a problem for me and happens even with a SINGLE 
working timeslot. But the behaviour of that problem is 
really strange. 

When I do connect from the chan_ss7 box to the endpoint over 
g711 codec - everything is fine.

When I do connect from the chan_ss7 box with IPP g729 or 
g723 codec with remote endpoint (over satellite link, 
600+ms) - sound is glitchy and there is an audio lost

When I do connect from the chan_ss7 box with g711 codec to 
ANOTHER asterisk box via SIP, which THAN transcodes g711 to 
IPP g729 or g723 codec with remote endpoint (over satellite 
link, 600+ms) - sound is glitchy and there is an audio 
lost!!!

When I do connect from the chan_ss7 box with g711 codec to a 
COMMERCIAL MVTS softswitch via SIP, which THAN transcodes 
g711 to it's own builtin g729 or g723 codec with remote 
endpoint (over satellite link, 600+ms) - sound is OK and 
there is NO AUDIOLOST!

So, I can't even guess why does that happen.

The error happens in the code which uses write() to the 
zaptel fd. Than write() returns EAGAIN and resource 
temporarily unavalable and that error happens. But 
considering the conditions given above - that is strange 
and I could only guess that there is some global 
desyncronization...

Regards,
Anton


On 20 March 2006 14:12, Kai Militzer wrote:
> Hello Anton,
>
> Anton wrote:
> > Did you ever experienced an AUDIO LOST problem?
> > (resource temporary unavailable) in write to zaptel
> > device?
>
> Not that I am aware off. How do you realize you have this
> problem? Is it hearable in calls in the audio stream or
> do you just see messages in the logs? I have a really,
> really low call volume on my machine, so the problem
> might not happen to me ...
>
> Best regards,
> Kai