I am afraid that we're going to sacrifice the "multiprotocol" aspect of Asterisk if we put DNS support in the dial plan. You have to be able to say
"call this URI, use these SIP servers for the call, time out a transaction with this time" Which is a very SIP-specific thing to do. We will have to implement a sip-dial() function in the dialplan with these specific arguments to get it right and let other people use the multiprotocol dial as before. And doing this just because some people have broken SRV records doesn't really indicate to me a good reason to have a broken SRV implementation in Asterisk. I like the comparision to broken MX records. If they don't want to receive calls and don't listen when you mail their sipmaster, well then. It's like forgetting to configure your MSN/DIDs when configuring ISDN. Broken. Kevin's code is a very good first step. Now we need to fix the SIP channel so that we cache these records, stay on the choosen record until it breaks, then pick the next one until that breaks and so on. I just wish Kevin's airplanes got delayed so he could focus more on coding... He he he. Sorry Kevin. /O _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
