hey list,
i currently test on my asterisk 1.6 box on on pc with private ip
and tested kamailio 3.1.3/3.1.4 + rtpproxy 1.2.1/1.2.0/1.1 on another pc with public ip and private ip.
everything installed successfully. i follow this tutorial realtime intergration with asterisk
http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb
my config.
rtpproxy -l publicip -s udp:127.0.0.1:7722 -u user
#!define WITH_NAT
modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722")
modparam("nathelper", "sipping_from", "sip:pinger@publicip")
nat_uac_test("19")
# uncomment next line to do SIP NAT pinging
setbflag(FLB_NATSIPPING);
is my config correct?
may i know the best version work out from the box?
please advice..
thanks in adv. :)
Hi,
did you also set
#!define WITH_NAT
at the beginning of your config? That should do the job.
Carsten
2011/6/24 MingHon gminghon@gmail.com:
hey list,
i currently test on my asterisk 1.6 box on on pc with private ip and tested kamailio 3.1.3/3.1.4 + rtpproxy 1.2.1/1.2.0/1.1 on another pc with public ip and private ip. everything installed successfully. i follow this tutorial realtime intergration with asterisk http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb
my config. rtpproxy -l publicip -s udp:127.0.0.1:7722 -u user
#!define WITH_NAT modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") modparam("nathelper", "sipping_from", "sip:pinger@publicip") nat_uac_test("19") # uncomment next line to do SIP NAT pinging setbflag(FLB_NATSIPPING); is my config correct? may i know the best version work out from the box? please advice.. thanks in adv. :) -- Regards,
MingHon
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
Hi,
Yup i did define #!define WITH_NAT at the beginning of config file. it doesnt work.
been struggle for a month hope you can help.
and also in the cfg im listening to both iface.
listen=public ip
listen=192.168.2.3 [kamailio ip]
Regards,
MingHon
Hi,
i should check your firewall settings.
Regards,
Stefano Larosa.
Da: sr-users-bounces@lists.sip-router.org [mailto:sr-users-bounces@lists.sip-router.org] Per conto di MingHon Inviato: venerdì 24 giugno 2011 17.56 A: sr-users@lists.sip-router.org Oggetto: Re: [SR-Users] rtpproxy and kamailio doesnt work out from the box.
Hi,
Yup i did define #!define WITH_NAT at the beginning of config file. it doesnt work.
been struggle for a month hope you can help.
and also in the cfg im listening to both iface.
listen=public ip
listen=192.168.2.3 [kamailio ip]
Regards,
MingHon
Hi,
which firewall do you mean? the uac firewall or the kamailio firewall?
kamailio doesnt have firewall. kamailio and rtpproxy in centos 5.4 firewall and selinux disabled.
eth0 is configured for ppp0e and eth1 is the private ip and
echo "1" > /proc/sys/net/ipv4/ip_forward
iptables -t nat -A POSTROUTING -o ppp0 -j MASQUERADE
and for asterisk is in fedora14 firewall and selinux also disabled.
for uac the firewall also disabled in the router. (dlink dir-615).
what else do i need to check? pls adv..
Hi,
i registered 3 uac behind same nat successfully but when i try to call each other i didnt get any audio. but if i use uac 102 and 103 to call into the voicemail i heard the audio but not for 101. kamailio is listening 60.48.218.61 and 192.168.2.3 rtpproxy is running. asterisk is at 192.168.2.23.
here is my ul show.
AOR:: 102 Contact:: sip:102@175.136.221.60:5062 Q= Expires:: 3110 Callid:: 721498432@175.136.221.60 Cseq:: 2 User-agent:: T22 7.3.0.50 Received:: sip:175.136.221.60:1024 State:: CS_SYNC Flags:: 0 Cflag:: 192 Socket:: udp:60.48.218.61:5060 Methods:: 16383 AOR:: 103 Contact:: sip:103@175.136.221.60:5062 Q= Expires:: 3114 Callid:: 1499738216@175.136.221.60 Cseq:: 2 User-agent:: Yealink SIP-T18 18.0.0.70 Received:: sip:175.136.221.60:1025 State:: CS_SYNC Flags:: 0 Cflag:: 192 Socket:: udp:60.48.218.61:5060 Methods:: 16383 AOR:: 101 Contact:: sip:101@175.136.221.60:5062 Q= Expires:: 3097 Callid:: 166053301@175.136.221.60 Cseq:: 2 User-agent:: T20 9.41.0.80 State:: CS_SYNC Flags:: 0 Cflag:: 0 Socket:: udp:60.48.218.61:5060 Methods:: 16383
and may i know why uac 101 did not have the received: field?
please some one could give a hand on this? the audio really cant get thru i really have no idea.
thank you
Hi,
i fixed the audio issue for 102 to 103 vice versa.
by fixing the canreinvite in asterisk.
from uac the rtp packet will route to kamailio den forward to asterisk.
can we bypass the rtp packet going to asterisk?
and here is the update for uac 101 issue.
when 101 call to voicemail or 102/103 there is no audio.
in wireshark i saw 101 send rtp packet to a private ip belong to asterisk.
but if 102/103 call to 101 both uac got audio.
i realize this is because 101 is the first uac registered before 102/103 and because it did not have the received: field in ul show.
please adv.
Try to run rtpproxy on private ip not on local 127.0.0.1
On Tue, Jun 28, 2011 at 11:23 AM, MingHon gminghon@gmail.com wrote:
Hi,
i fixed the audio issue for 102 to 103 vice versa.
by fixing the canreinvite in asterisk.
from uac the rtp packet will route to kamailio den forward to asterisk.
can we bypass the rtp packet going to asterisk?
and here is the update for uac 101 issue.
when 101 call to voicemail or 102/103 there is no audio.
in wireshark i saw 101 send rtp packet to a private ip belong to asterisk.
but if 102/103 call to 101 both uac got audio.
i realize this is because 101 is the first uac registered before 102/103 and because it did not have the received: field in ul show.
please adv.
-- Regards,
MingHon
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
Hi,
Thanks for the suggestion. i tried running rtpproxy on private.
rtpproxy -l public_ip -s udp:192.168.2.3:7722 -u user
and in kamailio cfg.
modparam("rtpproxy", "rtpproxy_sock", "udp:192.168.2.3:7722")
but still the same issue..
thanks.
Hi List,
i tcpdump on my kamailio server.
when uac registered with source port 5060
the uac will remain Cflag:: 0 and did not have the Received: field.
i also tried some iphone sip apps.
some uac app will register with src port 5060 and some will not.
anyone have solution for this?
or how to prevent uac register with src port 5060 ?
thanks!
-
Carsten Bock -
Chandrakant Solanki -
MingHon -
Stefano Larosa