[Asterisk-Users] Re: 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
> 

If I call from SIP to PSTN, at the beginning of the call
(1 second) after getting phone at the PSTN side I hear
voice at the SIP side. After this 1 second I don't hear
anything in SIP phone (at the PSTN phone everything is OK).

Nobody has had any problems like me?

Bartek.