asterisk users - Jan 2005 - I Don't Want Asterisk in the Media Path

Hi everybody.

I'm trying to find a way to connect two (or more) extensions directly
without
being kept in the middle during the conversation but it won't happen.

The purpose here is to have asterisk running on a low bandwidth (128Kbps)
internet connection just as some kind of a proxy between some ip phones with
high speed (10Mbps) internet connections.

SER is not an option, for now. I'm planning to have Asterisk and SER working
together in a near future.

I read a few documents (and comments) on the web regarding the reinvite option,
but I wasn't succesfull.

In my scenario, both ends have the same configuration: G.711 codec, no nat,
non-private ip address, canreinvite=yes.

Any suggestions or comments would be greatly appreciated. At this point, even
jokes...

Thanks in advance,

-Dhennys

"Dhennys Pestana" <list@pestana.com.br> writes:
> I'm trying to find a way to connect two (or more) extensions directly
without
> being kept in the middle during the conversation but it won't happen.
> [...]
> In my scenario, both ends have the same configuration: G.711 codec, no nat,
> non-private ip address, canreinvite=yes.
There are certain options to the Dial() command that will block
reinvites.  Perhaps you're using one or more of them?

-tih
-- 
Tom Ivar Helbekkmo, Senior System Administrator, EUnet Norway Hosting
www.eunet.no  T +47-22092958 M +47-93013940 F +47-22092901 FWD 484145

Dhennys Pestana wrote:
> I'm trying to find a way to connect two (or more) extensions directly
without
> being kept in the middle during the conversation but it won't happen.
Asterisk will always stay in the SIP signaling path.  It can get out of 
the RTP path (only way to really see this is using something like 
tcpdump since sip show channels shows the signaling not the RTP path). 
Asterisk CANNOT get out of the RTP path if you are using the "t" or
"T"
option to dial (maybe other options too) or if the codec for the two 
legs of the call are different.