On 02/22/2013 11:03 PM, Tod Hansmann wrote: > Where > FreeSWITCH gets into trouble is their community is much more "rtfm" and > less hand-holdy than Asterisk, though several of the community are more > than friendly, and they REALLY know their stuff for the most part.
Several years ago I had occasion to ask a few questions on the asterisk IRC channel and I found at least one of their main "experts" to be extremely RTFM-ish. He kept telling me I didn't know what I was talking about, when in fact it was he who didn't understand what I was asking. When I pushed back a bit he got all angry and basically said what I wanted to do was stupid anyways. Really soured me to asterisk. I've since learned that Asterisk really sucks at what I wanted to do. Basically I wanted to have a dialplan that could take a number, say BYU phone numbers, and turn them into url-based addresses ([email protected]) but pass them through a sip provider trunk instead of talking to sip.byu.edu directly. Unfortunately Asterisk just cannot deal with urls in their dialplans. It's a major weakness in asterisk, though an understandable one since telephones have number pads only typically. Anyay on IRC I was told to just use sip.byu.edu as a trunk and places the calls directly (since sip.byu.edu knows what to do with the numbers). The folks on IRC berated me for even wanting to do what I needed to do. Just fix my stupid firewall he said (which was outside my control). Of course if BYU actually registered their numbers with the ENUM service (DNS for SIP), then I could have simply used, say, sipgate.com as a trunk and they could use ENUM to figure out a no-cost dialing path. But since BYU doesn't (or hadn't at the time) I was kind of stuck. /* PLUG: http://plug.org, #utah on irc.freenode.net Unsubscribe: http://plug.org/mailman/options/plug Don't fear the penguin. */
