On the Asterisk side your firewall shall allow UDP port 5060 for SIP and some UDP ports for RTP (default 10000-20000 can be changed at /etc/asterisk/rtp.conf). Your sip.conf shall have Qualify=yes and Nat=yes.
On the telephone side, as long as your firewall allows outgoing traffic on 5060 and on the RTP ports (I believe this is default on these small routers) there will be no problems. Outgoing traffic on 123 will be nice also to have your phone clock updated. Rgds, Renato On Thu, 16 Sep 2004 18:45:09 -0400 (EDT), Mark Phillips <[EMAIL PROTECTED]> wrote: > Hi Folks, > > Anyone know how to make a grandstream phone work against a * server when > it is behind a cheap linksys type firewall? I have no control over the > firewall but am allowed to go anywhere I want. > > On the * end of the link there is another linksys type firewall which I do > control. What I don't have or particularly want is is a STUN server > running on my * server behind the firewall. > > Any ideas? > > Thanks > > -- > Mark Phillips, G7LTT/KC2ENI > Randolph, NJ > http://www.g7ltt.com/ > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
