Problems getting audio when call from outside mi LAN.
Both peers in my LAN perfect.
One peer in my LAN, and another outside in the cellphone company
internet; the one outside don't delivery any packet to the one inside my
LAN Them can see each other and try to talk but the packets sended to
the other party get lost. No one receive anything.
Restarting services for rtpproxy and kamailio after every try.
Here is my configuration on /etc/default/rtpproxy file:
USER=kamailio
GROUP=kamailio
CONTROL_SOCK="unix:/var/run/rtpproxy/rtpproxy.sock"
EXTRA_OPTS="-l 72.28.208.16 -s udp:localhost:7722"
Here my configuration on kamailio.cfg file corresponding with NAT:
#!ifdef WITH_NAT
# ----- rtpproxy params -----
#modparam("rtpproxy", "rtpproxy_sock",
"udp:127.0.0.1:7722")
modparam("rtpproxy", "rtpproxy_sock",
"unix:/var/run/rtpproxy/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@kamailio.org")
# params needed for NAT traversal in other modules
modparam("nathelper|registrar", "received_avp",
"$avp(RECEIVED)")
modparam("usrloc", "nat_bflag", FLB_NATB)
#!endif
Ports opened in my router for Kamailio:
udp:35000-65365, udp-tcp:5060, 5061
I know this is personal information but this server is a case study. I'm
Trying to learn from you all and if it's possible contribute in the
future with some knowledge.
Two users for you to test:
user name: support
domain:
uscallfree.com
password: support
Not TLS yet
user name: master
domain:
uscallfree.com
password: master
Not TLS yet
If one of you wants to try what happen when calls from one peer use that
one above, my user name always active is admin(a)uscallfree.com. I know
it's not working but maybe helps to determine what it's happening.
admin(a)uscallfree.com active 24 hrs. in my nexus 4 native SIP profile
configuration.
Thank for the help provided here in the mailing list. You all rocks