Ok thanks,
All fine except the rtpproxy_manage function is just out in the open and since you're in bridging mode you need to realize that your kamailio may receive calls from the WAN Interface or from the Asterisks on LAN interface. So if you're to bridge the RTPs at the proxy then how can Kamailio detect that which combination of flags to use in rtpproxy_manage() function?
As an example if you replace your rtpproxy_manage(); with rtpproxy_manage("ei"); and make a call from SIP user to 777 in asterisk dialplan where 777 is echo()
exten => 777,1,Answer() same => n,Echo()
Take a sip trace now and see that you get the desired LAN IP of kamailio coming in on Asterisk ? revert the "ei" to "ie" if not. This simple test should get you a two way audio established for any call from Internet to the Asterisk via Kamailio.
The Next step is to figure out that your call is coming from Asterisk and wants to go out from kamailio's WAN interface. use inverse flag combination here and let the call go out.
I hope it works for you now.
-- Sammy
On Tue, Aug 6, 2013 at 4:06 AM, Alexandr Usov blessendor@gmail.com wrote:
ps -ef | grep rtpproxy kamailio 15853 1 0 12:41 ? 00:00:01 /usr/sbin/rtpproxy -u kamailio -l 1.1.1.1 2.2.2.2 -s unix:/var/run/rtpproxy.sock
netstat -pln|grep rtpproxy unix 2 [ ACC ] STREAM LISTENING 238561268 15853/rtpproxy /var/run/rtpproxy.sock
I'm trying rtpproxy_manage("ei"); rtpproxy_manage("ie"); rtpproxy_manage(); - the same if rtpproxy in bridged mode.
Config cuts:
#!define WITH_NAT ... #!ifdef WITH_NAT # ----- rtpproxy params ----- # modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:12221") modparam("rtpproxy", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock")
# ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@sip.myrealhostdomain.com ")
# params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif
....
# Handle SIP registrations route[REGISTRAR] { if (is_method("REGISTER")) { if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); # uncomment next line to do SIP NAT pinging setbflag(FLB_NATSIPPING); } if (!save("location")) sl_reply_error();
exit; }}
...
# Caller NAT detection route route[NATDETECT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { fix_nated_contact(); } setflag(FLT_NATS); } #!endif return;
}
# RTPProxy control route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB);
} } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; rtpproxy_manage(); if (is_request()) { if (!has_totag()) { add_rr_param(";nat=yes"); } } if (is_reply()) { if(isbflagset(FLB_NATB)) { fix_nated_contact(); } }#!endif return; }
2013/8/6 SamyGo govoiper@gmail.com
Please check the rtpproxy function and paste the way it is written in your configuration file. Share the output of "ps -ef | grep rtpproxy" and "netstat -pln|grep rtpproxy"
-- Sammy
On Tue, Aug 6, 2013 at 3:55 AM, Alexandr Usov blessendor@gmail.comwrote:
Note: Asrterisk LAN IP real 192.168.144.101 but must be 2.2.2.101 in this described network (I am missed to change before copy-pasting here).
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users