[Asterisk-Users] SIP conections, with RTP not going trough Asterisk

[Peter Bowyer]

On 17/04/06, Tiago Stein D`Agostini <tiago at khomp.com.br> wrote:
> So, is there any other option that prevents that from happening?
> Something that I might have turned on  and makes Dial work  trough
> asterisk? I already even removed asterisk completelyu from system and
> reinstalled it to be fresh new... still all RTP goes trough Asterisk
> machine. And the server really can't handle many connections this way.

What options are you using? Post an extract of your dialplan and sip.conf.

And how are you determining that the RTP is going through Asterisk?

Peter

--
Peter Bowyer
Email: peter at bowyer.org