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/[EMAIL PROTECTED]/[EMAIL PROTECTED],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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