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.