Google Voice ------ FreePBX ------ CUCM

Hello all,

I have been working with FreePBX for sometime now and I have been having a roller coaster of a time getting it to full work with Google Voice. I have the google voice module installed and it has made the Google Voice trunk for me and I have a SIP trunk going to my cisco call manager that is hosting the phones.

For awhile I was able to successfully make outbound calls from the cisco IP phones through GV, but suddenly it has halted. Unfortunatly I cannot tell you if it was something I did in the process of making the inbound calls work but I have reverted to earlier notes of when it did work and nothing. The PBX Call logs simply states the call is exiting the PBX and then has NO ANSWER.

I have never been able to successfully make inbound calls. When I call the GV number, it rings, connects, and the simply drops the call. The call is received by the PBX server and then “tdials” the SIP trunk to the CUCM. I am having a hard time understanding what digits are being transferred to the CUCM for further call manipulations. In the call logs it states that the call is ANSWERED. I have included my server’s version numbers and my configuration. If anyone can tell me how to attach a file I can show you a quick sample of a incoming call from the server.

PBX in a Flash Version = Running on VMWARE
│ FreePBX Version = │
│ Running Asterisk Version = │
│ Asterisk Source Version = │
│ Dahdi Source Version = │
│ Libpri Source Version = 1.4.12 │
│ IP Address = on eth0 │
│ Operating System = CentOS release 5.6 (Final) │
│ Kernel Version = 2.6.18-238.9.1.el5 - 64 Bit │

Dial Plan Configuration:
Outbound Routes:
Inbound Route:

Thanks to anyone that may help!

Regarding Outgoing Calls, there was a problem started on Friday, October 14. The discussion and fix are here:—-maybe

Hope that fix the outgoing calls.

Thank you very much, the outgoing is working again. Now any advice on the incoming??

I am curious as to what you are using this for?

Just testing purposes, I am a college student trying to learn the most I can.

Can you make regular calls between FreePBX and Call Manager?

Make sure that the Calling Search Space on the SIP trunk in the Cisco Call Manager is set correctly.

I have got the Trunk setup with a Calling Search Space named Generated_CSS_I_E and have added 3 phones to the same Call Search Space.

I have read in this post:
To make a ring group and I have done so. I can make the PBX server once again call the tdial and sends it down the SIP trunk and then the call is terminated with no answer on the cell phone end.

I am still up for any ideas and thankful for the ones so far.