Asterisk and OpenSer

Asterisk Asterisk Support
1

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…

2

Which one has the non-standard port number?

3

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) */
#listen=udp:10.0.0.10:5060
listen=172.16.52.143

/* port to listen to

    • can be specified more than once if needed to listen on many ports */
      port=5060

#!ifdef WITH_TLS
enable_tls=yes
#!endif

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

These parameters can be modified runtime via RPC interface

- see the documentation of ‘cfg_rpc’ module.

Format: group.id = value ‘desc’ description

Access: $sel(cfg_get.group.id) or @cfg_get.group.id

#!ifdef WITH_PSTN

PSTN GW Routing

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

pstn.gw_ip = “10.0.0.101” desc “My PSTN GW Address”

- by default is empty to avoid misrouting

pstn.gw_ip = “” desc “PSTN GW Address”
#!endif

#!ifdef WITH_ASTERISK
asterisk.bindip = “172.16.52.143” desc "Asterisk IP Address"
asterisk.bindport = “5080” desc "Asterisk Port"
kamailio.bindip = “172.16.52.143” desc "Kamailio IP Address"
kamailio.bindport = “5060” desc “Kamailio Port”
#!endif

4

and of course this is asterisk sip.conf:

openser*CLI> sip show settings

Global Settings:

UDP SIP Port: 5080
UDP Bindaddress: 0.0.0.0
TCP SIP Port: Disabled
TLS SIP Port: Disabled
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