Tod, Just a couple of thoughts...
1) If your astLinux box is behind NAT, then your sip.conf variable externip=<your_public_ip> must match your public WAN address. With a dual-WAN router I'm not sure how you are handling this. 2) If your ATA's sip.conf context contains: canreinvite=no should guarantee the RTP voice path will go through your astLinux box. Lonnie On Nov 24, 2008, at 5:05 PM, Tod Fitch wrote: > Situation: Have two ISP links each with a static IP. Current router > does not understand SIP, so ports 5060 and 10000-20000 are forwarded > to Asterisk. On my v1.2 system Asterisk always acts as a B2BUA and > everything is fine. > > Moved to AstLinux v0.6.1 with Asterisk v1.4 running on a Soekris > Net5501 and while calls to/from registered ITSP work, calls to ENUM > destinations fail as the Asterisk attempts to route RTP from the > phone to the ENUM destination. At least my firewall status page show > the ATA sending RTP packets to the ENUM destination so I think this > is what is happening. (I have been trying to figure out how to keep > Asterisk in the path but the NAT and reinvite settings for outbound > appear to have no effect in this case). > > So. Looked at changing router to an embedded box running pfSense, > but apparently the embedded version of pfSense does not support > packages and it would require the siproxd package to be any better > than my current machine. > > Next option: Enable the Arno firewall in AstLinux and put on the > phones into their own network that has to go through the AstLinux > box. That should keep the dual-Wan router happy. It looks like this > should work (EXT_IF set to current LAN, INT_IF set to new VoIP > specific network, set FULL_ACCESS_HOSTS to allow management from the > LAN, and some revised cabling and new Ethernet switches). This will > be my fall back position. > > (Sorry for the long winded introduction...) > > However it would be cleaner if I replaced the current firewall/ > router with the AstLinux box. Has anyone done this with multiple WAN > interfaces? What issues did you run into? Was the Arno firewall > multiroute plug-in sufficient? Do you need to use the sip-voip plug- > in? > > --Tod > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's > challenge > Build the coolest Linux based applications with Moblin SDK & win > great prizes > Grand prize is a trip for two to an Open Source event anywhere in > the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/_______________________________________________ > Astlinux-users mailing list > [email protected] > https://lists.sourceforge.net/lists/listinfo/astlinux-users > > Donations to support AstLinux are graciously accepted via PayPal to [EMAIL > PROTECTED] > . ------------------------------------------------------------------------- This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK & win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100&url=/ _______________________________________________ Astlinux-users mailing list [email protected] https://lists.sourceforge.net/lists/listinfo/astlinux-users Donations to support AstLinux are graciously accepted via PayPal to [EMAIL PROTECTED]
