Hi there
I have set up freepbx with SipStation and work good im all softphone
when It come to polycom ip 650 it has a problem when it to call it(extension example 200) when try to dial 300 the call goes
Scenario 2:300 to 200
then extension goes automatically to busy state
can you post the console output from asterisk -r -vvvv? log in to your asterisk through ssh, at the prompt type: asterisk -r -vvvv make a call from 300 to 200 and post what you see.
Dial(“SIP/800-00000048”, “SIP/200,20,800I”) in new stack[2016-12-26 17:58:16] WARNING[14382][C-000000f2]: app_dial.c:2437 dial_exec_full: Unable to create channel of type ‘SIP’ (cause 20 - Subscriber absent)
this is what I got but i think my phone is registered
try: sip show peers
200/200 122.x.x.x D Yes Yes A 2543 UNREACHABLE
figuring out why this is unreachable
it seems that your phone is accessing asterisk through internet, when you call 300 from 200 does the voice work from both sides? check your nat settings in the extension’s advanced options. Try setting it to “route”. Also check that your phone is set to register itself with asterisk. If the phone does not register it can still make calls but cannot receive any. What model phone do you use?
polycom SoundPoint IP 650
Do the two way voice test, check your phone web interface to make sure the extension is set to register with asterisk, verify your nat settings.
Also make sure that under settings->advanced sip settings your external ip is correctly set
other polycom phones are working correctly except this polycom
external IP is perfectly ok 
The path through your NAT/firewall times out sooner than your SIP registration. You need to lower the SIP registration timer on the device, enable the device’s NAT keepalive functions, or try a stateful transport such as TCP.