[asterisk-dev] One way audio problems

[Asbjørn Zweidorff Kjær]

Tilghman Lesher wrote:
> On Friday 17 November 2006 08:52, Martin Vít wrote:
>> What about reinvite=no or somthing like this to avoid native
>> transfer?
> 
> There is no such parameter as "reinvite=no".  The parameter you
> are referring to is named "canreinvite=no".  If you are not precise,
> you cause confusion and frustration for those trying to solve issues.
> 

Just tried setting notransfer=yes to avoid asterisk getting out of the
media path and it still happens (canreinvite is a sip.conf setting, its
iax.conf equivalent is notransfer). Ive tried it with both yes and no
for settings (of course changed on both ends of the link)

Any other ideas at what i might try to avoid this?

-Asbjoern


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