[Asterisk-Users] re: help with voicepulse connect IAX2

[yair hakak]

hello,
after playing with an asterisk configuration for voip for a few weeks i'm 
trying to get outbound dialing with voicepulse going - i've cut down the 
asterisk to a very minimal install (1 SIP client) to try to localize the 
problem. The SIP client works fine (SIP and * on the same NAT) and could 
access the demo from samples before i removed it,  and can call itself  - so 
i am pretty convinced the SIP setup is OK.

This is the error message:
Jan 29 12:21:54 NOTICE[262161]: app_dial.c:527 dial_exec: Unable to create 
channel of type 'IAX2'
when i try to call the PSTN from the SIP device.
i've tried the wiki, the handbook, the voicepulse site, and all sorts of 
other sites, and nothing helps. i also downloaded and compiled the code 
today (jan 29) and that didn't help either. if anyone has ideas i would be 
eternally grateful - this is driving me crazy.

thanks-
yair

p.s. i am using the right login and password; not the ones from the website, 
and i know my account at voicepulse works because i can connect direct 
through a SIP client.  it seems to be a specifically IAX2 problem.

here are my files

sip.conf
;
; SIP Configuration for Asterisk
;
[general]
port = 5060                     ; Port to bind to
disallow=all                    ; Disallow all codecs
allow=ulaw                      ; Allow codecs in order of preference
allow=gsm

[yairphone]
type=friend
insecure=no
username=yairphone
secret=yairphone
host=dynamic
dtmfmode=inband
callerID = "Yair Hakak"
nat=true

extensions.conf
[general]
;
static=yes
writeprotect=no


[default]

exten => _1NXXNXXXXXX,1,Dial(IAX2/myUserName at voicepulse/${EXTEN}@VPWS,20)
exten => 8665,1,Dial(SIP/yairphone,20)

iax.conf
[general]
port=5036
disallow=all
allow=ulaw

jitterbuffer=no

[voicepulse]
context = VPWS
secret=mypassword
auth=md5
type=friend
host=gw5.voicepulse.com

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