Config Files For New PIAF Users (Cut & Paste)

[jay.johnson]

When I started learning PBX-In-A-Flash/FreePBX back in September of last year, I told myself that once I understood the "basics" fairly well, I'd share what I knew

Attached, you'll find the four basic configs that when added under the "Tools--->Config Edit" GUI menu in PBX-In-A-Flash/FreePBX will consistently/successfully nullify the following known issues (at least from what "I've" experienced)

:

1) Dropped calls

2) One-Way-Audio

3) Ringing not being heard when a number is dialed

4) Phone call does not disconnect after hanging up

5) IVR menu does not respond when number is pressed (this is more of an issue with dtmfmode under "peer details")

The other half of this equation is ensuring your "peer details" (under the "Trunk" heading") are accurate. Below are a couple of config examples from the SIP providers I use (feel free to cut/paste/substitute your information where needed). Also, if you use the 4 attached text file configs, delete any commentary inside of parentheses as well as any arrows that are pointing to something that's illustrating a point/idea:

--------------------------------------------------------------

1) SipGate (USA)

username=XXXXXXXeX
type=peer
secret=password
qualify=yes
outboundproxy=proxy.live.sipgate.com
nat=yes
insecure=invite,port
host=sipgate.com
fromuser=xxxxxxxx
fromdomain=sipgate.com
dtmfmode=rfc2833 (Use rfc2833 when using IVR)
disallow=all
canreinvite=no
call-limit=5
allow=ulaw,alaw.g729,gsm

SipGate Registration String:

XXXXXXXeX[email protected]/XXXXXXXeX

NOTE 1:"XXXXXXXeX" = your SIP-ID assigned to you from SipGate

NOTE 2:"Password" = your SIP-Password assigned to you from SipGate

----------------------------------------------------------------

2) CallCentric

username=1777XXXXXXX
type=peer
secret=password
qualify=yes
nat=yes
insecure=invite,port
host=callcentric.com
fromuser=1777XXXXXXX
fromdomain=callcentric.com
dtmfmode=rfc2833 (Use rfc2833 when using IVR)
disallow=all
context=from-trunk
canreinvite=no
authuser=1777XXXXXXX
allow=ulaw,alaw,g729,gsm

CallCentric Registration String:

1777XXXXXXX[email protected]/1777XXXXXXX

NOTE 1:"1777XXXXXXX" = your SIP-ID assigned to you from CallCentric

NOTE 2:"Password" = your SIP-Password assigned to you from CallCentric when you first create your account

----------------------------------------------------------------

Additionally, here're the two basic "getting started" links from both providers I started out with:

1) http://www.callcentric.com/support/device/freepbx

2) http://www.sipgate.com/faq/article/394/How_do_I_configure_Asterisk

In the end, the two examples are what "I" modified down to what I use daily without too many issues (if any). Sure, everyone has a different approach, but I'm finally free of the "newbie" problems I mentioned previously (i.e., drop outs, one-way-audio, etc...)

Let me know if this is of use to anyone...

Jay