One Way Audio

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!

Although it is probably not your problem (for which I’d need to see the SIP dialogue), that version of Asterisk contains no string to match against the “very” in insecure=very, and, given you have no secret defined, I can’t think why you would have needed it before it changed its name.

Also, to interface with a CUCM trunk you do not need to use friend, peer is enough, and a single sip.conf entry is perfectly sufficient. There is no need for fromdomain to be anything other than that of the Asterisk box, and I would be slightly confused that setting it the same as the Cisco might actually confuse it. I think these are FreePBX artifacts.

canreinvite is deprecated in that version, but still recognized.

As well as the SIP trace, the Google protocol trace is probably also needed, but, personally, I don’t know how to interpret that, so you had better hope that the problem is on the SIP side.