[Asterisk-Users] Extensions SIP problems.

[David Uzzell]

I am playing with the dialplan to get it working and I have a challange 
with this error. I can't find what it means on the wiki :(

Any sugestions would be helpful at being able to forward it to the SIP 
phone if it is online and avaliable but then let that fail and drop into 
voicemail if it is not online or is busy.

cheers

David

-- Executing Dial("IAX2/firefly at 89280250/3", "SIP/800|5") in new stack
Dec 21 00:15:57 NOTICE[3922]: app_dial.c:800 dial_exec: Unable to create 
channel of type 'SIP' (cause 3)
   == Everyone is busy/congested at this time
     -- Executing WaitExten("IAX2/firefly at 89280250/3", "") in new stack


The Extensions.conf file for that section is

exten => s,1,Wait,1
exten => s,n,Answer
exten => s,n,DigitTimeout,3
exten => s,n,ResponseTimeout,5
exten => s,n,Dial(SIP/800,5)
exten => s,n,Waitexten
exten => s,n,Playback,voicemail/default/801/unavail
exten => s,n,Voicemail,801
exten => s,n,Goto,t|1


and I have in sip.conf

[800]
type=friend
regexten=800
username=800
secret=password
callerid=800
host=dynamic
;dtmfmode=inband
mailbox=800
nat=yes
canreinvite=no
qualify=yes
disallow=all
allow=gsm
allow=speex
allow=ilbc
allow=ulaw
allow=alaw