Asterisk as NOT a proxy


#1

Hi everybody … A few days agos a have installed a asterisk sip server im my network. It works perfectly, except that my sip server in acting as a sip proxy. I mean, all “conversation” cross through my sip, and its not P2P.
For example:
client1 has ip 10.0.0.1
client2 has ip 192.168.0.1
sip server has ip 10.0.0.5.

All conversation is between client1 <-> sip_server <-> client2,
But i want client1 <-> client2 (they talking direclly). Is that possible ?

Before somebody answer, two things: first, sorry about my “english” (and don’t know speek very well); and second, sorry for my silly question (maybe stupid for you all).


#2

Asterisk will attempt to do this by itself by default when possible. The call control layer will pass through Asterisk for billing purposes, but if it determines two phones can talk to each other it will send a REINVITE message to let the phones directly transmit audio to each other.

Depending on your sip.conf settings, this may be disabled. If you’ve set canreinvite=no either globally or for a specific phone, Asterisk will not issue the REINVITE and will act as a bridge between the two endpoints.

Direct connections area also generally not possible if your phones are behind NAT, since there’s no way for outside phones to establish communications with them through the firewall.