On Wed, May 6, 2009 at 12:48 PM, [email protected] <[email protected]> wrote: > Can anybody provide any suggestions to help debug this? If I'm unable to > isolate/resolve the problem then its likely we'll have to drop the > Asterisk solution and I've already grown rather attached to it.
I have a number of ideas of what could be happening, and most involve routing issues over your VPN, or your VPN dropping packets. Here's a suggestion: * put another asterisk server on the remote side, and have the two asterisk servers do SIP or IAX trunks back and forth. If you don't want to invest in a server, at least pull an old computer off the curb and do some tests using that computer. If your phones come unregistered but your SIP trunk is fine, change your branch office phones to register to their local asterisk instead, and set your remote server accordingly. You might need to do some prefixing, or redirects, or other tricks to make the trunking transparent to the users if you don't want to reassign extensions. _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
