On Thu, 2003-08-14 at 22:41, John Todd wrote: > Since nobody else took the hint, I submitted it as a feature request for SIP. > > http://bugs.digium.com/bug_view_page.php?bug_id=0000104 > > Personally, this is not high on my "I'd love to see this fixed" list. > However, many others here are less fortunate from a network > perspective, and you're stuck behind a cable modem or DSL router.
As one of those who didn't take the hint I'll put forward my defence. John, I could not have written the additional information in any way with the detail you did. I think there are new users like me who have never even seen a Cisco in operation, and certainly don't yet understand the intricacies and problems associated with SIP. Perhaps this is a problem with *, virtually out of the box you find yourself talking to someone at Digium :). Except for a minor annoyance with sometimes not dropping incoming analog calls in France correctly (workaround restrict the length of the recorded message, who wants to hear their life story anyway?) within a day you have the basis of a working system. You can then start expanding the system, and then you hit the wall, SIP. The list grows STUN, Partysip, SER, Sarp, FCP, etc... With a completely operational Linux firewall system, providing you with all you need, and more, from a pensioned off PC, the best advice people can give you is buy the XYZ SIP aware firewall. If XYZ can do it why can't we. I am in no way a telecoms expert, but even at my advanced age I can still write C, even Cobol, but first I need an idea of what is needed, a spec or something. -- Dave Cotton <[EMAIL PROTECTED]> _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users