Hi Gilles, > If someone has a working configuration where... > 1) Asterisk and some users are on a private LAN behind a NAT firewall > 2) some roadwarriors, behind their own NAT firewall, are allowed to > register with Asterisk, and make/receive calls just like they were in > the office > 3) the NAT firewall protecting the Asterisk server has SIP and > RTP/RTCP ports mapped, while the NAT firewall protecting the remote > user has its ports open dynamically using STUN
I have (almost) that and it is working fine. One user needed to use a different port than 5060 because his modem really loves to interfere with packets on 5060. Probably because their ISP can also provide a SIP/phone line and the ISP modem is assuming that all packets on 5060 are for that phone line even though it is not enabled. I don't use STUN anywhere. Anyway, in all cases it has worked as soon as the phone was able to register on my server (that was usually the hard part!). In sip.conf I have added for all the remote users the setting canreinvite=no. The downside to that setting is that Asterisk is always in the audio path. For my situation that does not really matter. Best regards, Jeroen Eeuwes -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
