Asterisk and OpenSer

Hi guys, I am tryng to understand how Kamailio ( OpenSer) works. I tried to implement both asterisk and kamailio on same box. I am able to install them, and use kamailio as register. Now I cannot understand how to make able phones calling each other. Does anyone tried this solution? I mean kamailio as SIP PROXY SERVER;REGISTER ecc… and asterisk as simple VOICE Server…
Thanks in advance for any advise or share experinces…

Which one has the non-standard port number?

I setted asterisk to listen to port 5080 and Kamailio to port 5060.
This is a part of the Kamailio.cfg code:

/* uncomment and configure the following line if you want Kamailio to
bind on a specific interface/port/proto (default bind on all available) */

/* port to listen to

    • can be specified more than once if needed to listen on many ports */

#!ifdef WITH_TLS

####### Custom Parameters #########

These parameters can be modified runtime via RPC interface

- see the documentation of ‘cfg_rpc’ module.

Format: = value ‘desc’ description

Access: $sel( or

#!ifdef WITH_PSTN

PSTN GW Routing

- pstn.gw_ip: valid IP or hostname as string value, example:

pstn.gw_ip = “” desc “My PSTN GW Address”

- by default is empty to avoid misrouting

pstn.gw_ip = “” desc “PSTN GW Address”

asterisk.bindip = “” desc "Asterisk IP Address"
asterisk.bindport = “5080” desc "Asterisk Port"
kamailio.bindip = “” desc "Kamailio IP Address"
kamailio.bindport = “5060” desc “Kamailio Port”

and of course this is asterisk sip.conf:

openser*CLI> sip show settings

Global Settings:

UDP SIP Port: 5080
UDP Bindaddress:
TCP SIP Port: Disabled
TLS SIP Port: Disabled