asterisk users - Dec 2004 - Lets try this again then! Q: SIP error from dialplan I suspect!

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@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@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

is this current cvs or something? It looks completely abnormal for stable..

seems you are doing a lot of extra stuff you don't need to.. I'd see if 
just this works for you..

exten => 800,1,Dial(SIP/800,60)
exten => 800,2,VoiceMail(800)

also.. why disallow all and then allow most everything? seems like you 
are trying to over think things.. no offense.
why not slim down your peer entry a bit?
ie:

[800]
type=friend
username=800
secret=password
callerid=800
host=dynamic
dtmfmode=inband
mailbox=800
nat=yes
canreinvite=no


David Uzzell wrote:
>
> 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@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@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
>
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