[asterisk-users] Configuring Softphone

[Danny Nicholas]

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Gary Kuznitz 
Sent: Wednesday, December 08, 2010 1:27 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Configuring Softphone

The phone is finally registering.   That's great.

I'm trying to understand what all these lines in Extensions.conf are
defining.
I can't make heads or tails them.  I have been reading the manual 
AsteriskManualTheFutureOfTelephony2ndEdition.

I'm currently getting an error when placing a call on the cmd line saying:
NOTICE[5630]: chan_sip.c:14383 handle_request_invite: Call from 'Gary' to 
extension '91AreaCodePhone#' rejected because extension not found.
 

What I have in Extensions.conf is:
[gary-incomming]
exten => 1001,1,Dial(IAX2/gogh)
exten => 1001,2,HangUp()
exten => 120,1,Dial(SIP/Gary)
exten => Gary,1,Goto(120,1)
exten => i,1,Playback(invalid)
exten => i,2,Goto(s,1)
exten => s,1,Wait(1)
exten => s,2,Answer()
exten => s,3,NoOp(${CALLERID})
exten => s,4,NoOp(${CALLERIDNUM})
exten => s,5,NoOp(${CALLERIDNAME})
exten => s,6,Wait(4)
exten => s,7,Playback(vm-goodbye)
exten => s,8,Wait(2)
exten => s,9,HangUp() 

What I have in Sip.conf is:
[authentication]

[general]
context = default
allowoverlap = no
bindport = 5060
bindaddr = 0.0.0.0
srvlookup = yes
limitonpeers = yes
allowguest=no
nat=yes         

[Gary]
type = friend
username = Gary
callerid = 120
secret = password
host = dynamic
defaultip = dynamic
context = gary-incomming
dtmfmode = rfc2833
allow=all  

Frustrated,

Gary

Without any other comment, you need 
exten => _91.,1,Dial(DAHDI/g1/${EXTEN})
in the gary-incomming context.

As defined now, Gary can 
#1 answer a call
#2 call IAX/gogh using 1001