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Sip Config for Aretta CBeyond and Voiceflex with Asterisk

Since Version 1.8 in Asterisk we have seen some issues with DID calls from some suppliers.

The tell tail sign is that even though you have an inbound route that matches the DID it will still say in the verbose screen that nothing matched it in the inbound context, For example:-

Call from 'USERNAME' (XXX.XX.XXX.XX:5060) to extension '01234123412' rejected because extension not found in context 'from‐trunk'

and if you do “dialplan show 01234123412@from-trunk” sure enough there is one.

After much searching and experimentation below is a working freepbx config that has been tested with 1.8 and 11 and proves to be working with the suppliers above.

OUTBOUND

[peername]
username=USERNAME
type=peer
trustrpid=yes
sendrpid=yes
secret=PASSWORD
qualify=no
outboundproxy=sip.hostname.com
nat=yes
insecure=very
host=sip.hostname.com
fromdomain=sip.hostname.com
dtmfmode=auto
disallow=all
context=from-trunk
canreinvite=no
allow=ulaw
allow=alaw

INBOUND

[username]
type=peer
host=sip.hostname.com
dtmfmode=auto
disallow=all
context=from-trunk
canreinvite=no
allow=ulaw
allow=alaw

;registration string
USERNAME:PASSWORD@peername/USERNAME