27 May
2009
27 May
'09
12:01 p.m.
2009/5/27 Chandrakant Solanki <solanki.chandrakant(a)gmail.com>om>:
Hi
I am using kamailio 1.5.0 and asterisk 1.6.0.5
Phone is register successfully and incoming/outgoing call is working fine...
But followings are problem I have faced...
1] I don't hear anything, the voice is not being transmitted ... however
rtpproxy is running.
2] When I hang up the X-Lite call on the SIP phone, the regular phone does
not get disconnected.
You are not fixing correctly the NAT issues (Contact header and/or SDP
media address) in both requests and responses.
I suggest you to do a SIP capture (with ngrep) and check the IP's in
Contact headers and SDP after passing through kamailio. Are they
fixed?
Also, if you are using Asterisk there are TOO MUCH possibilities: Have
you using Asterisk comedia mode ("nat=yes") for peers? do you allow
direct RTP between peers ("canreinvite=yes")?
However, using Asterisk to manage a call I think it's not a good idea
to use RtpProxy (it's not neede at all most probably!).
Below is link for kamailio configuration file...
http://en.pastebin.ca/1435982
Yes, there is... but there are more factors (Asterisk conf, network
topology and so...). Impossible to examine all of them (at least by
free XD).
--
Iñaki Baz Castillo
<ibc(a)aliax.net>