[Asterisk-Users] Re: Codecs.

[Rich Adamson]

> > > Hi all i have some problems with my pbx and asterisk codecs.
> > > 
> > > if i use g711u or g711a codecs. the line never hangup. and the origin
> > > and destination are connected until i restart my pbx or asterisk
> > > 
> > > But if i use GSM all work fine.
> > > 
> > > is possible to solve this problem? or use only gsm codec?
> > 
> 
> > Yes, its possible to solve the problem.
>
> can you explain how?

Not without you providing at least "something" to give us a clue what it
is that you've programmed into your system. 

How about if you give us some clue as to which version of * you're
using, what type of phones are associated with "origin" and "destination",
if these are sip phones what do your sip.conf definitions look like,
what does the appropriate sections of extensions.conf look like, and
any other configuration pieces that might pertain to whatever it is
that you've implemented. Your posting implies there might be more than
one * system involved and possibly even iax trunking, etc.