Hi
I have setup an Asterisk 1.4.0 pbx currently with support for only SIP to try out VoIP.
I have all the internal calls configured to my needs but I am having trouble with the outbound calls, I have registered with freecall.com to test VoIP, and configured it to use SIP to connect. This link shows some settings which I used in sip.conf
Here is sip.conf:
[general]
context=default
allowoverlap=no
bindport=5060
bindaddr=0.0.0.0
[l0]
host=sip.voiparound.com
qualify=20000
secret=**********
type=peer
username=[my username]
[301]
type=friend
secret=**********
qualify=yes
nat=no
host=dynamic
canreinvite=no
context=internal
I have X-lite configured as a softphone which connects to the Asterisk PBX.
Here is extensions.conf
[incoming]
exten => s,1,Answer()
exten => s,2,Background(vm-enter-num-to-call)
exten => _3XX,1,Goto(internal,${EXTEN},1)
exten => 700,1,VoiceMailMain()
exten => t,1,Playback(vm-goodbye)
exten => t,2,Hangup()
[internal]
exten => 600,1,Goto(incoming,s,1)
exten => _3XX,1,Dial(SIP/${EXTEN})
exten => _3XX,2,Playback(vm-nobodyavail)
exten => _3XX,3,VoiceMail(${EXTEN}@default)
exten => _3XX,4,Hangup()
exten => i,1,Background(pbx-invalid)
exten => 700,1,VoiceMailMain()
include => outbound-local
[outbound-local]
exten => _1NXXXXXX,1,Dial(SIP/00649${EXTEN:1}@l0)
What is happening is after dialling 1 630****, it displays “Calling…” for a while and says “Call failed: Forbidden”.
In the asterisk console with vvvv set, it shows this:
– Executing [1630****@internal:1] Dial(“SIP/301-08793a98”, “SIP/00649630****@l0”) in new stack
– Called 00649630****@l0
[Feb 16 14:19:40] WARNING[24359]: chan_sip.c:11731 handle_response_invite: Received response: “Forbidden” from ‘“Mr.Bobster” sip:301@192.168.1.2;tag=as493bba32’
– SIP/l0-08797a00 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
However when using the software provided by freecall, it works normally.
I am new to Asterisk and was following the “Asterisk The future of telephony” pdf book when setting up the extensions.
I would like to ask how I can set it up so that these local calls can be dialed.
Thanks in Advance
Jason
Edit: Added Asterisk Version in use (1.4.0)
