CallCentric trunk setup with Asterisk/FreePBX
Here is my CallCentric configuration for FreePBX.
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Trunk Name: CallCentric
PEER Details:
username=1777XXXXXXX
type=peer
secret=PASSWORD
qualify=yes
nat=no
insecure=very
host=callcentric.com
fromuser=1777XXXXXXX
fromdomain=callcentric.com
dtmfmode=rfc2833
disallow=all
context=custom-get-did-from-sip
canreinvite=yes
allow=ulaw
Register String:
1777XXXXXXX:PASSWORD@callcentric.com/1777XXXXXXX
PEER Details:
username=1777XXXXXXX
type=peer
secret=PASSWORD
qualify=yes
nat=no
insecure=very
host=callcentric.com
fromuser=1777XXXXXXX
fromdomain=callcentric.com
dtmfmode=rfc2833
disallow=all
context=custom-get-did-from-sip
canreinvite=yes
allow=ulaw
Register String:
1777XXXXXXX:PASSWORD@callcentric.com/1777XXXXXXX
Please note: the above number starting 1777 is your account number and not you DID number
You also need to add 2 lines to one of the configuration files to correctly extract the DID number from incoming calls.
Edit /etc/asterisk/extensions_custom.conf
Add the following lines
[custom-get-did-from-sip]
exten => _.,1,Goto(from-trunk,${CUT(CUT(SIP_HEADER(To),@,1),:,2)},1)
exten => _.,1,Goto(from-trunk,${CUT(CUT(SIP_HEADER(To),@,1),:,2)},1)
Then restart Asterisk
You should now be able to create Inbound Routes based on you CallCentric DID numbers
Related posts:
wipmonkey:
No support for g729(a) ??
24 July 2009, 11:03 pmmatt:
Callcentric do support G729 so you can change the allow= line and add it there. You’ll need a codec/license for g729 through for your Asterisk box unless you just want to do pass-thru (which means you can’t transcode or do DTMF)
24 July 2009, 11:08 pmvfyzskcsarjnh:
To-based routing is wrong.
29 November 2009, 4:43 pmhttp://www.kempgen.net/voip/sip-request-uri-vs-to-header-routing.html
matt:
While it may not be RFC compliant unfortunately it’s necessary to route based on the format provided by the ITSP
29 November 2009, 5:01 pm