I have asterisk 1.8.4.1 installed in our server along with freepbx.My problem is that whenever an incoming calls are made,the call comes in to asterisk but it finds all the extension lines as congested and hangs up.I had an inbound route to transfer the incoming calls on my pri number 555 to a sip extension registered with asterisk.But whenever i call the lines are found as congested.
these are my configs:
dahdi-channel.conf:http://pastebin.com/neznAxFh
chan_dahdi.conf:http://pastebin.com/XFqCiHi1
inbound route:http://imgur.com/BXfqx
extension:http://imgur.com/rVV78
from-pstn context in freepbx was used
debug info for call :http://pastebin.com/CykAK5bk
The calls come in to asterisk for sure,but for some reason it finds the line of the sip extension congested.Anyone help plz…cant figure out the problem.