hi but i don't think IAX2 is good, because with IAX2 RTP packets goes via IAX servers as mini packets not directly from one client to other client so for a big implementation it may consume more bandwith then that of a SIP solution rest is up to the user... ----- Original Message ----- From: "Wilson Pickett" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <[email protected]> Sent: Tuesday, August 02, 2005 11:07 PM Subject: Re: [Asterisk-Users] port forwarding ip to ip sip calls
> I've got two pa1688 phones that I want to set up to communicate between > branch offices without a gatekeeper. Both phones will be behind a > firewall and I want to use port forwarding so the phones can communicate. Are you using these phones with SIP? Why not try IAX2? > I tested the phones behind a firewall on the same network segment and > there were no problems at all using sip. However, I then moved the > phones into situ and port forwarded udp on 5060 and 10000 - 20000 at > both branch offices firewalls. I set the rcp port to 10000 and the sip > port to 5060. The phones were able to ring each other, however, there > was no sound on both ends. > > Can some one please tell me which ports I have to open in order to make > communications between the two branch offices using these phones. Or > share a config or suggest another protocol so I can make this happen. Check for nat=yes and canreinvite=no in sip.conf _______________________________________________ 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
