[Asterisk-Users] Perhaps something obvious?

[Matt Herzog]

On Wed, Dec 29, 2004 at 12:21:51PM -0500, Matt Herzog wrote:
> I am a VoicePulse.com user although I have never been able to connect.
> I have no dialtone nor can I determine if I have been authenticated.
> Do I need to configure for sip? I was told I did not need SIP. 
> Voicepulse does support sip . . . 
> 
And I forgot to ask, do I need to forward the ports to the Asterisk server
or the SPA device? 

Here are my configs:


; Sample /etc/asterisk/iax.conf downloaded from VoicePulse and edited
; by MSH subsequently.
; Created September 1, 2004

[general]
port=5036
tos=lowdelay
jitterbuffer=no

; ---------------------------------------------------------
; The following codecs are support by the VoicePulse 
; Connect! service:
; ---------------------------------------------------------
disallow=all
allow=ulaw
allow=ilbc
allow=gsm
allow=adpcm
allow=alaw
;allow=g726     ; g726 is NOT supported as of 10/1/2004,
                ; but is coming soon.

; ---------------------------------------------------------
; This is how you tell VoicePulse Connect! gateways where 
; to send your incoming calls.  The 10 characters before 
; the ":" are your VoicePulse Connect! gateway login and 
; the 10 characters after the colon are your Connect! 
; gateway password.  You can find this information by 
; logging into your VoicePulse Connect! account at 
; http://connect.voicepulse.com and clicking on Devices.
; ---------------------------------------------------------
; 
; The entire "register =>" line below should be on one line
; (with no carriage returns in the middle):

register => nkv87PBo43:Huk44Hwx85 at 66.234.228.170
register => nkv87PBo43:Huk44Hwx85 at gwiax-in-01.voicepulse.com

; ---------------------------------------------------------
; We use RSA keys for authentication purposes.  If you 
; haven't already saved the VoicePulse public key, you can 
; get it by doing the following from a shell prompt: 
;
;  cd /var/lib/asterisk/keys
;  wget http://connect.voicepulse.com/keys/voicepulse01.pub
; ( I installed their pub key. -- MSH ) 
; This is a guest user to catch all unauthenticated calls
; 
[guest]
type=user
context=guest

;
; This is the VoicePulse user for incoming calls to your
; Asterisk server:
;
[voicepulse-in-01]     ; <-- Name must be [voicepulse-in-01]
type=user
context=incoming	       ; <-- Should match the context you 
auth=rsa
inkeys=voicepulse01

; This is a test user.  You can use Dan Toma's DIAX Software
; Phone to test your Asterisk configuration.  Set the
; following in the DIAX > Config > Registration menu option:
; 
;   Server: <your Asterisk server IP address>
;   Username: diax
;   Password: diaxpassword
; 
; You can get DIAX at: 
; http://www.laser.com/dante/diax/diax.html
; I eschew Windows.

;[diax]
;type=friend
;context=outgoing
;auth=md5
;secret=diaxpassword
;notransfer=1
;host=dynamic
;allow=gsm


; Sample /etc/asterisk/extensions.conf
; Created September 1, 2004
; Edited by MSH thereafter.
; =========================================================
; QUICKSTART WITH VOICEPULSE CONNECT! SERVICE:
; * Login to your VoicePulse Connect! account at:
;   http://connect.voicepulse.com/
; * Go to the Devices tab and note your device login and
;   password
; * Replace MY_DEVICE_LOGIN and MY_DEVICE_PASSWORD in the
;   "exten => " statements below with your device login
;   and password. (Lines 81-82)
; * If you DO NOT have a phone number from VoicePulse
;   Connect!, comment out the following lines by placing a
;   semicolon ";" at the beginning:
;   - The entire "[arbitrary-name]" context (lines 43-48)
;   - The entire "[testdtmf]" context (lines 54-60)
; =========================================================
[general]
static=yes
writeprotect=no

[globals]

; [arbitrary-name] is the context referred to by the 
; [voicepulse-in-01] user in iax.conf.  This is where your 
; custom incoming call processing should go.
; For sample purposes, this section will read back the 
; dialed number and then test DTMF by reading back each 
; digit pressed by the caller.
;
; I don't unserstand this part at all. Do I put my phone number here?
; -- MSH
; ---------------------------------------------------------
[incoming]  ; <-- Should match the context you have 
                  ;     under [voicepulse-in-01] in iax.conf
exten => _NXXNXXXXXX,1,Playback(beep)
exten => _NXXNXXXXXX,2,SayDigits(${EXTEN})
exten => _NXXNXXXXXX,3,Goto(testdtmf|s|1)
; 
; This context is used by the sample [arbitrary-name]
; context above to read back each digit you press.
; 
[testdtmf]
exten => s,1,Background(beep)
exten => s,2,ResponseTimeout(60)
exten => _x,1,SayDigits(${EXTEN})
exten => _x,2,Goto(testdtmf|s|1)
exten => i,1,Goto(testdtmf|s|1)
exten => t,1,Hangup

; ---------------------------------------------------------
; This context is used to send all outgoing calls to the
; VoicePulse Connect! service for connection to the PSTN.

; Asterisk will attempt to dial out through gwiaxt01 first.
; If there is a problem, it will attempt to dial out
; through gwiaxt02.

; YOU MUST HAVE BOTH LINES FOR OUTGOING CALL REDUNDANCY!
;
; ---------------------------------------------------------

; 
; There should be TWO lines after [outgoing], each beginning
; with "exten =>".  Please check to make sure copying or 
; cutting & pasting this sample did not break the lines into 
; more than TWO exten lines.
; 
[outgoing]
exten => 4042146081,1,Dial(IAX2/NKV99PBo43:hUK44hWX73 at gwiaxt01.voicepulse.com/${EXTEN})
exten => 4042146081,2,Dial(IAX2/NKV99PBo43:hUK44hWX73 at gwiaxt02.voicepulse.com/${EXTEN})