I am using the Asterisk 1.6.2 and I setup the two SIP extensions let say 2001 and 2002. When I am calling from 2002 to 2001 then 2001 getting the incoming call ring. But after the answered the call there is no voice.
And one more issue is that when I am calling from 2001 to 2002, then i get the Busy message. But the 2001 phone is free.
I have done the following settings in sip.conf
[2001]
type = friend
user = 2001
secret = 2001
fromuser = 2001
context = default
nat = yes
qualify = yes
host=dynamic
disallow = all
allow = all
[2002]
type = friend
user = 2002
secret = 2002
fromuser = 2002
context = default
nat = yes
qualify = yes
host=dynamic
disallow = all
allow = all
Can you please let me know what I am missing in configuration.
Voice quality problems are due to network or CPU overloads, or due to high scheduling latencies as the result of trying to run on a virtual machine without very careful setup and control of conflicting loads.