[asterisk-users] extensions.conf / test DID

[Jacob.E.Miles at L-3Com.com]

I am trying to make sure my DID and SIP account details are working
properly and engaging the extensions.conf and dial plan.

I have a successful SIP session registered:

Connected to Asterisk 11.3.0 currently running on Asterisk (pid = 922)
Asterisk*CLI> sip show registry
Host                                    dnsmgr Username       Refresh
State                Reg.Time
sip3.voipvoip.com:5060                  N      1112530146         105
Registered           Mon, 08 Apr 2013 06:02:09
1 SIP registrations.
Asterisk*CLI>

Here is the dial plan:
[incoming]
exten => 17036361355,1,Playback(beep)
exten => 17036361355,2,SayDigits(${EXTEN})
exten => 17036361355,3,Goto(testdtmf|s|1
;Ring on Elle  mobile phone.
;exten => s,1,Answer()
;exten => s,n,Dial(SIP/17037171234,150,r,t,)


[general]
register =>1112530146:albany!@#123 at sip3.voipvoip.com/1112530146
registertimeout=20
context=incoming
allowoverlap=no
bindport=5060
bindaddr=192.168.1.10
srvlookup=no
;context=incoming

; The SIP provider
[voipvoip.com]
canreinvite=no
username=1112530146
fromuser=1112530146
secret=albany!@#123
context=incoming
type=friend
fromdomain=sip3 at voipvoip.com
host=69.90.209.57
dtmfmode=rfc2833
disallow=all
allow=alaw
allow=ulaw
nat=force_rport
insecure=port,invite

Thoughts please?    I think something to do w/ "incoming" is incorrect.



 

[incoming]
exten => 17036361355,1,Playback(beep)
exten => 17036361355,2,SayDigits(${EXTEN})
exten => 17036361355,3,Goto(testdtmf|s|1
;Ring on Elle  mobile phone.
;exten => s,1,Answer()
;exten => s,n,Dial(SIP/17037171234,150,r,t,)

 

As well doesn't the Goto need to closing ")"?

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130408/24956ec0/attachment.htm>