[asterisk-users] Call hangup after 60seconds

[Il Neofita]

Hi,
I have a client (xlite) connected to my server, on the server I have
type=friend and siptimeout=60, canreinvite=yes and dial with tT option, the
server is listening on port 5060.
However, xlite is connect to a router where the port 5060 is blocked,
therefore, I am using 5065 and I have an iptables rule to transfer the
incoming packet from 5065 to 5060,
I cannot use the port 5065 since some ATA the do not allow the change of the
port.
When I am calling with xlite the call endup after 60seconds, but in the
60seconds I can talk.
Now if I am setting the client (in the sip.conf) in peer everything is
working.

Someone can explain to me why? What I am doing wrong?

Thank you
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070924/cd323ff2/attachment.htm