One Way Audio

Asterisk Asterisk Support
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!

2

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.