Sip does not register

I am having a pretty weird problem here today. I signedup for Voicepulse and ever since I installed everythinig I cannot make any sip calls.
I am running asterisk 1.4 beta … Its a test box!
this is what happens when i make a call

Sep 25 12:05:27] NOTICE[19670]: chan_sip.c:13178 handle_request_invite: Unable to create/find SIP channel for this INVITE
[Sep 25 12:05:47] WARNING[19670]: chan_sip.c:1847 retrans_pkt: Maximum retries exceeded on transmission 2179419f904c7af9@10.X.X.X for seqno 53073 (Critical Response)

here is my sip.conf

[general]
context=default ; Default context for incoming calls
allow=all

[204]
; Turn off silence suppression in X-Lite (“Transmit Silence”=YES)!
; Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed
type=friend
regexten=204 ; When they register, create extension 1234
callerid=<204>
host=dynamic ; This device needs to register
secret=xxx
canreinvite=no ; Typically set to NO if behind NAT
;disallow=all
allow=gsm ; GSM consumes far less bandwidth than ulaw
allow=ulaw
allow=alaw
allow=h263
context=default
;mailbox=1234@default,1233@default ; Subscribe to status of multiple mailboxes

[203]
; Turn off silence suppression in X-Lite (“Transmit Silence”=YES)!
; Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed
type=friend
regexten=203 ; When they register, create extension 1234
callerid=<203>
host=dynamic ; This device needs to register
secret=xxxx
;canreinvite=no ; Typically set to NO if behind NAT
;disallow=all
allow=gsm ; GSM consumes far less bandwidth than ulaw
allow=ulaw
allow=alaw
allow=h263
allow=all
context=default
;mailbox=1234@default,1233@default ; Subscribe to status of multiple mailboxes

any ideas? thanks.

I read your sip.conf set up. I am not using Voicepulse. However,
“Unable to create/find SIP channel for this invite” indicates Asterisk
is looking for a trunk and registry which it cannot find. Anyway, here
is mine through Sipnumber as my Voip provider.

sip.conf

under the "register section’

register => username:password@voipprovider.com
my username is my phone number but use whatever Voicepulse gives you.

[sipnumber]
type=friend
host=sipnumber.net
username=my phone number
fromuser=my phone number
fromdomain=sipnumber.net
secret=???
insecure=invite,port
nat=yes (my box is behind a NAT Firewall)
qualify=yes ( Same as above, this addresses the NAT isssue )
canreinvite=no
dtmfmode=rfc2833
disallow=all
allow=ulaw
allow=alaw
allow=gsm
context=from-sipnumber

[7010] (my extension number)
type=friend
username=7010
secret=???
authuser=7010 (authuser is only used by this extension, not for X-Lite)
host=dynamic
context=local
end
Then I have a [from-sipnumber] and a [sipnumber-out]
with a defined dialplan in extensions.conf and in the [default]
section I added “include=sipnumber-out” (no quotes).In the [local] section I commented "include => default. This works seemless. Also, I noticed, you have a semi-colon (:wink: before
some of your comments in sip.conf. Watch this because Asterisk
will interpret this as a “comment out” meaning it will bypass the
comment.

If you need the extensions.conf full comments (dial plan), let me know
and I will post them.

Hope this helps !