Ok so some good news, incoming calls via the sip trunk work, but we still can't make outgoing calls, just a busy tone is made after trying to dial the number.
Thanks for all the feedback J Derek Bartolo Technical Support Comsource Technologies 3585 Laird Drive Unit 13 Mississauga Ontario [email protected] <mailto:[email protected]> 905-608-1400 ext 240 From: [email protected] [mailto:[email protected]] On Behalf Of Tim Byng Sent: Thursday, July 30, 2009 3:05 PM To: Scott Lawrence Cc: Bob Anderson; [email protected] Subject: Re: [sipx-users] auto attendant | SIP Trunks On Thu, Jul 30, 2009 at 2:40 PM, Scott Lawrence < [email protected]> wrote: On Thu, 2009-07-30 at 13:25 -0400, Bob Anderson wrote: > > Also has anyone set up a sip trunk from unlimitel using SIPX bridge http://list.sipfoundry.org/archive/sipx-users/msg15054.html I was never able to get Unlimitel working perfectly with sipXecs 4.0.1. Standard inbound and outbound calls worked well, but there was an issue with call forwarding to external numbers (the call would be dropped). I believe Ranga is currently testing a fix that may resolve this problem, so it may work in a future version. BTW, I think Unlimitel does hosted NAT compensation, so you may want to set "Use public address for call setup" to false and "Method to use for RTP keepalive" to "Use empty packet". Although it works with "Use public address for call setup" set to true, I think this will improve the quality, as you don't want both Unlimitel and sipXecs trying to do NAT compensation (someone please correct me if this is wrong).
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