This is my first post in the forum, so if I am doing anything wrong, or could be doing it better please let me know.
(tried to follow these (http://support.freepbx.org/forum/freepbx/installation/so-you-have-a-problem-and-want-help) suggestions)
I recently (Dec 2010) setup a FreePBX server with 5 Polycom 500 SIP phones
I used a PBXIn a Flash Distro (Purple) 1.7.9
FreePBX ver: 22.214.171.124
Asterisk ver: 1.8.0
CentOS ver: 5.5
We set it up on a DSL line, and (with significant help from this forum, and others)got everything working in a way we were satisfied with.
However, our fax machine was on the same line as the DSL and caused problems.
Last week we changed from the DSL to a fibernet line and had to change our router
from the DSL to an EnGenius ESR9850 (incoming ‘fibernet’ connection is technically Dynamic, but the ISP guarantees the IP address so it works like a static).
Our SIP Trunks are provided by SIPStation and we used the SIPStation module to configure everything initially.
Since we have updated the router, we have not been able to get calling set up the way we want.
Outgoing calls will ring the intended phone, but when the call answers there is no audio – either direction.
When that phone tries to call back, the call goes straight to voicemail.
When any of the 5 phones tries to call any other internal extension we get a busy signal.
When we try to dial in for voicemail (*98) we get nothing.
It seems reasonable to me that this is some how a router issue (though I have since turned off the router’s firewall), but I don’t know
how to troubleshoot this issue any further.
I have tried looking through an Asterisk log during an incoming call, but don’t really know what I am looking at/for, and I am stumped.
Any help on this issue, or thoughts about additional troubleshooting steps I can take would be greatly appreciated.
PS - I have generated Asterisk outputs for: an attempt to dial voicemail (*98) with no answer, and an internal dial (from ext 200 to ext 300).