Hi
I need help on NAT with Kamailio...
My Network Details are as follows...
SIP Phone Registration ADSL Router Kamailio Asterisk
172.18.100.73/255.255.0.0 192.168.1.1/255.255.255.0 192.168.1.68/255.255.255.0 192.168.1.70/255.255.255.0 172.18.100.72/255.255.0.0 GW: 172.18.100.72 (Virtual IP 172.18.100.68) (Virtual IP 172.18.100.70)
Router Type: AirLink 101 802.11 Token Ring
Where Virtual IP 172.18.100.68 and 172.18.100.70 set virtually on ADSL Router.. and set routing on it..
So when I ping 172.18.100.70 it will go through 192.168.1.1
I am able to call, receive means able to receive OK, INVITE and REGISTER method of kamailio ... but When i try to Hangup call, it can't hangup at both side.. I have to manually cut phone at the other end..
I have not set any rule on Router.
*Scenario...*
1] I have two user. 1212@domain.com [172.18.100.73] and 2121@domain.com[172.18.100.72] and register successfully 2] Call from 1212@domain.com [172.18.100.73] to 2121@domain.com[172.18.100.72].. 3] Call is connected successfully 4] While hangup from 1212@domain.com [172.18.100.73] it will take time and at the other end, I have to manually hangup call..
Please help me out..
Do a SIP capture (ngrep -d any -W byline -T -P '' port 5060) and examine where the packets are flowing...
Do it on both so you can see where the BYE gets lost.
On Thu, Jun 11, 2009 at 11:46 AM, Saúl Ibarrasaghul@gmail.com wrote:
Do a SIP capture (ngrep -d any -W byline -T -P '' port 5060) and examine where the packets are flowing...
-- Saúl -- "Nunca subestimes el ancho de banda de un camión lleno de disketes."