Reduce load on Asterisk

I have a simple setup with an Asterisk server (V1.2.8 ), 2 IAX phones and 2 SIP phones connected to a switch.

What I saw

When I initiate a call from one phone to the other I see on the sniffer that all the traffic is going to and from the Asterisk server.

Suppose the amount of phones increases, so the load on the Asterisk server will increase too.


Is there a configuration parameter or a method to make the Asterisk server only doing the call setup ( like Cisco call manager ) and leave the conversation between the phones ?

Thanks in advance !

If “conversation” means the rtp stream you can try set canreinvite=yes, in sip.conf, for every phone.


Same result !

If the conversation is between the IAX phone and SIP phone, then Asterisk has to stay in the media path to do protocol conversion. Theres no way around this.
Asterisk by default will try to make call flow effiecient by letting TRP flow direct between endpoints, but if protocol or codec conversion is needed, RTP will flow into Asterisk.