I am having difficulties getting my PBX (PIAF 18.104.22.168 running Asterisk 1.4.33) to talk to the Windstream/Nuvox SIP server. They have given me very little information on how to make it work and so far their tech support (network guys) have been unwilling to escalate me to their SIP people.
Here is all they have given me:
: G711ul, 20ms sampling, silence suppress (19 preferred, 13 supported)
DTMF Relay (RFC 2833):
Nuvox's Signaling and Media Ports:
SIP Signaling port: 22.214.171.124 port 5060
DNS Servers (Nuvox):
NTP Servers (Nuvox):
Nuvox expects to receive 10 digits from PBX from domestic local calls and 1+10 for long distance.
Nuvox will provide 10 digits back to the PBX for "DNIS" call routing.
No SIP Registration/Authentication is required.
NuVox will re-mark all traffic coming from your PBX with an IP Precedence of 3 for SIP Signaling and 5 for Media/RTP.
Long distance, 911, 411, International will follow normal dialing rules.
I found this forum post with some information on how to configure my trunk settings - I did it and the thing still won't work. http://fonality.com/trixbox/forums/trixbox-forums/open-discussion/sip-signaling-port-74223147141-port-5060-mediartp-74223147140
The trunk settings the above link suggested are this:
I used those (and many variations) with no luck.
We have confirmed in our edge router logs that we are sending traffic to the SIP server on port 5060 but don't get any traffic back.
The bottom line is this - I need to know what my trunk settings should be. Especially the Outgoing PEER details and the Incoming USER details sections. Thanks in advance for the help.