[Asterisk-Users] Lets try this again then! Q: SIP error from dialplan I suspect!

[Matt Hess]

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 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
>
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