I am trying to get a telephone line set up with the Google Voice Gateway found at: https://simonics.com/gvgw/. The problem is that I have no clue what to put in the PEER Details and USER Details of the FreePBX trunk. I am currently able to get it to register but, when I dial out all I get is “All Circuits are Busy Now” and when I dial in I get the Google Voicemail and the Asterisk log does not indicate that a call was ever received.
I have tried asking on the Google Voice Gateway forum but, it seems to be a bit inactive so, I am trying here.
Any and all help is thanked and please remember that I am not some linux/FreePBX/SIP guru, just a slightly knowledgeable nerd trying something new.
I haven’t played with it but from a quick search looks like the registration string should be simonicsusername:[email protected]/11 Digit Number
and
Trunk should have 1 in prepend and the NXXNXXXXXX in Dial Pattern
Outbound Route has 1 in prefix and NXXNXXXXXX
I have used the registration string that the website provides. Let me give you a few more details. The SIP port is 5070 and the RTP ports are 16384-32768. I have set the bindport in the Asterisk SIP Settings menus to 5070 and the RTP ports to those listed above. I have also completed the NAT setting section of that same menu and edited the sip_nat.conf file to include externip=108.XX.XXX.XXX
localnet=172.8.0.0/255.255.0.0
I have also followed the guide here: http://nerdvittles.com/?p=832
The FreePBX dashboard shows the trunk as being registered but, when I make an outbound call the “All Circuits are Busy Now” error plagues me and the CDR report shows the call as failing due to Congestion. Incoming calls do not even show up on the CDR report. I know the trunk works because I can use SIPDroid to log into it with my cell phone.
Like I said I have not played with it before, but what do you have for inbound and outbound routes?
http://wiki.freepbx.org/display/F2/Inbound+Routes
http://wiki.freepbx.org/display/F2/Outbound+Routes