Fongo Community Forums

Hello,

Could anyone please remind me the settings Freephoneline needs to work with Asterisk (FreePBX) ?

I tried to replicate whatever has been mentioned on this forum a few years ago, with no luck.

Thanks,

[general]
context=unauthenticated ; default context for incoming calls
allowguest=no ; disable unauthenticated calls
srvlookup=yes ; enabled DNS SRV record lookup on outbound calls
udpbindaddr=IP.ADD ; listen for UDP requests on all interfaces
tcpenable=no ; disable TCP support
register => 1234567890:SECERETPASSWORD@voip2.freephoneline.ca:5060
acl=tellme
useragent=CISCO-TEL

[office-phone](!) ; create a template for our devices
type=friend ; the channel driver will match on username first, IP second
context=LocalSets ; this is where calls from the device will enter the dialplan
host=dynamic ; the device will register with asterisk
nat=yes ; assume device is behind NAT
; *** NAT stands for Network Address Translation, which allows
; multiple internal devices to share an external IP address.
secret=VERYBIGSECRET ; a secure password for this device -- DON'T USE THIS PASSWORD!
dtmfmode=auto ; accept touch-tones from the devices, negotiated automatically
disallow=all ; reset which voice codecs this device will accept or offer
allow=ulaw ; which audio codecs to accept from, and request to, the device
allow=alaw ; in the order we prefer


This may be very late for your answer but I am sure that others will be interested. You need to setup SIP not PJSIP. Modify your sip.conf accordingly. Below is an example that works with the proper username and password.

[freephoneline]
type=peer
secret=SECRETPASSWORD
username=1234567890
host=voip2.freephoneline.ca
fromuser=1234567890
fromdomain=YOUR.IP.ADD.OR.YOURDOMAIN
canreinvite=no
insecure=invite,port
qualify=yes
nat=force_rport,comedia
context=from-sip ; this section will be defined in extensions.conf
deny=0.0.0.0/0.0.0.0
permit=162.213.111.22/255.255.255.255