We have a Centos 6.5 VPS running FreePBX. I’m brand new to FreePBX so please mind my learning curve
We have a trunk setup with a sip provider which forwards all calls to our FreePBX server. I also have an incoming route setup with a sip number which also works. I am able to make an incoming call and it is registered by Asterisk (I am able to get the ivr prompt). I then have 2 extensions setup (201, 202) and for each of these I have a Yealink W54P phone setup locally. The Yealink base has 2 accounts each registered to 201,202 above. I am able to make a phonecall outbound from the Yealink phones through my trunk. I am just not able to get incoming calls directed to the extensions. My current setup has incoming calls coming into our IVR messaging and via user keypress will then forward the call to the appropriate extension. The calls are immediately directed to voicemail.
Trunk Settings:
host=xxx
username=xxx
fromuser=xxx
secret=xxx
type=friend
sendrpid=yes
trustrpid=yes
disallow=all
allow=alaw&g722
qualify=yes
Asterisk output when we register the base accounts:
Connected to Asterisk 13.5.0 currently running on voice (pid = 1890)
-- Unregistered SIP '201'
-- Registered SIP '201' at 10.0.0.50:5062
When an incoming call is made to the inbound
[2015-08-24 13:34:40] WARNING[6159][C-00000011]: app_dial.c:2397 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
== Everyone is busy/congested at this time (1:0/0/1)
Output of sip show peers:
Name/username Host Dyn Forcerport Comedia ACL Port Status Description
201/201 10.0.0.50 D No No A 5062 UNREACHABLE
202/202 10.0.0.50 D No No A 5063 UNREACHABLE
268460001/268460001 195.35.115.119 No No 5060 OK (178 ms)
I hope the above will be suffice. Thank you!