Andreas Sikkema wrote: > > On 6/13/13 16:20 , Matthew J. Roth wrote: > > It's hard to be certain without seeing a full SIP trace, but I think the > > INVITE > > with the internal IP is actually a re-INVITE that Asterisk is sending to > > establish a native bridge between the SIP friend and the SIP gateway to PSTN > > converter. > > It's actually pretty easy. > > If an INVITE message has a tag parameter in both To and From headers, > it's a re-INVITE. If the To header doesn't have a tag parameter, it's an > initial INVITE.
Andreas, Thanks for the tip. That's a very useful bit of information to know. It also confirms that the INVITE in Mickael's original post [1] is a re-INVITE: From: <sip:xxxx at 109.69.217.6>;tag=as15b47581 To: "test" <sip:xxxx at 109.69.217.6>;tag=kp1VwHD80rA9MVdBjTF4jyFIaCkrJcjh Consequently, the advice in my response [2] about initially reconfiguring Asterisk with "directmedia=no" and then fine-tuning the NAT SUPPORT and MEDIA HANDLING settings is valid. [1] http://lists.digium.com/pipermail/asterisk-users/2013-June/279435.html [2] http://lists.digium.com/pipermail/asterisk-users/2013-June/279450.html Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer -- _____________________________________________________________________ -- 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