Make sure your dialplans are "enabled" and have a gateway assigned to
them.
On Jul 30, 2009, at 2:08 PM, Derek Bartolo wrote:
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]
905-608-1400 ext 240
<image001.jpg>
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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