One way audio

FreePBX Installation / Upgrade
1

I have searched this issue several times and although there are plenty of topics on this issue, the many I have looked at do not seem to match my topology or setup exactly and the fixes don’t work on mine.

Here is my topology:
Google Voice ------ Router ----- PBX (PIAF) ----- CUCM -----IP Phones

Currently I can call out using the IP phones to an external cell phone just fine, but incoming the ip phone rings and I pick up, but the external phone continues to ring and ring.

I am using the Google Voice Module that was added in and seems to be working. I have opened ports 5060 & 10000-20000 and pointed them to the PBX (which I don’t if this needs to be CUCM). This is my SIP trunk configuration:

VERSIONS:
Asterisk 1.8.4.1
FreePBX 2.9.0.7
PIAF 1.7.5.6
CUCM 8.03

PEER DETAILS:
host=
type=friend
qualify=yes
nat=no
insecure=very
fromdomain=
dtmf=rfc2833
disallow=all
context=from-internal
canreinvite=no

USER DETAILS:
type=friend
qualify=yes
nat=no
insecure=very
host=ip.address.of.CUCM
fromdomain=192.168.3.25
dtmf=rfc2833
disallow=all
context=from-internal
canreinvite=no
allow=ulaw

Please let me know if there is anymore information you need!
pamiller3
Newsterisk

Posts: 1
Joined: Fri Feb 03, 2012 5:43 pm

2

Well, you have partial PEER Details and that has to cause problems. Also, I realize that people are able to get Google voice working in some fashion, it is not intended to work like that and IMHO a moving target. Try a pay-as-you-go SIP trunk from Vitelity or les.net as an inexpensive way to work out the bugs in your setup and network.

Also, you mention ports forwarded… Was that TCP or UDP ? Needs to be UDP.

Lastly… CUCM ? Cisco Unified Call Manager ? So you are trying to use Asterisk as a proxy to get Google Voice to a Call Manager system which does not sound like a real good idea. Have you put an ext on the PIAF and see if it behaves them same ? Since CUCM is in the mix, you might mention which version and what IP Phones.

3

Could you elaborate on the partial PEER Details. I have seen many use PIAF as a SIP proxy for CUCM. I am shooting to use Google Voice because of the free service, I am a cheap college student. I have forwarded the UDP ports. And yes Cisco Call Manager, the version is 8.03 and the phones are 7975s, 7971s, CIPCs, & 7960s. Some of these are using SIP profiles and some are SCCP.

Please keep the suggestions coming.