* You are viewing the archive for July 7th, 2012

Rookie / Sip And Extensions

Sorry for blasting another desperate note but I am trying! I have changed the username and password and IP to protect my system.
But, the logic is unchanged. It is does not work! I simply want to dial the telephone number provided to me for my DID which corresponds with my
SIP info.
And, then it should connect and hit the “incoming” context and simply dial the 617 number. I am close but no cigar. Now I get a fast busy tone only.

What is missing or what is needed please?

extensions.conf
[globals]

;
;
[incoming]
;
;exten=> s,1,Goto(125010155_incoming)
;
;[125010155_incoming]
exten => s,1,Answer exten => s,n,Dial(SIP/16175551212)


sip.conf
[general]
;register => 125010155:funnytiger@sip3.voipvoip.com/125010155
register => 125010155@sip3.voipvoip.com:funnytiger@69.90.209.11
;
[incoming]
username5010155
type=peer secret=fnnytiger nat=ato insecure=invite,port hosti.90.209.11
fromdomaini.90.209.11
dtmfmode=rfc2833
context=incoming allow=g729
allow=ulaw allow=aaw allow=ilbc srvlookup=yes

Trixbox Or FreePBX Or Elastix Or PBX In A Flash

Thanks Tim.

One of the problem that I am facing is the complicated generated configuration for the FreePBX, is it the same thing in the Elastix?

To understand this complicated generated commands, is there a documentation to explain this for FreePBX or Elastix?


One of my friend told me that he installed (as I remember) FreePBX and there were already existed the TFTP files that the Cisco IP Phones is requesting (for sip or skiny) and already there were a TFTP server. Which module to do this?

Regards
Bilal

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