[Asterisk-Users] sip+oh323 - no voice at sip side

[bartek at datacomsa.pl]

On 26-07-2005 at 07:23:39PM +0200, bartek at datacomsa.pl wrote:
> Hello,
> I have something like this:
> SIPUSER <-sip-> ASTERISK <-oh323-> AUDIOCODEC <-e1-> PSTN
> 
> After calling from SIP to PSTN (and from PSTN to SIP too)
> I can't hear anything only in my SIPUSER. At the PSTN side everything is OK.
> 
> I have another network with another h323/sip (in the place of asterisk)
> and there everything is OK.
> 
> In AUDIOCODES logs I see that everything goes fine with asterisk, but SIPUSER
> can't hear the PSTN user.
> 

The problem was in oh323.conf:
fastStart=no (was yes)

Now everything goes fine.

Bartek.