Hi Lonnie, Thank you for your quick reply.
In my sip.conf I have: externip=primary_isp_ip ;external=secondary_isp_ipAnd I edit sip.conf to change which line is commented out if/when the primary fails. A bit ugly and I'd like to get away from it if possible.
I tried adding a "canreinvite=no" to both the [general] and the ATA specific contexts in sip.conf. It works for v1.2 but that was working anyway. For v1.4 it does not seem to do any good. I think that no invite was made to the ATA prior to establishing the connection with the ENUM destination. In this case Asterisk v1.4 seems to be acting more like a SIP proxy here than a B2BUA. I haven't yet tried putting an ANSWER in the dial macro ENUM to force an invite to the ATA before setting up with the final ENUM destination, that might make the "canreinvite=no" work. It might also create a problem if the ENUM fails and I want to then continue on with a try to the ITSP.
--Tod On Nov 24, 2008, at 3:33 PM, Lonnie Abelbeck wrote:
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 worldhttp://moblin-contest.org/redirect.php?banner_id=100&url=/ _______________________________________________ Astlinux-users mailing list [email protected] https://lists.sourceforge.net/lists/listinfo/astlinux-usersDonations to support AstLinux are graciously accepted via PayPal to [EMAIL PROTECTED] .
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