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

[David Uzzell]

Matt Hess wrote:
> is this current cvs or something? It looks completely abnormal for stable..
> 

Ah sorry it is CVS 12/12/04

> 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)

Cool thanks for that. It creates the same error. :(


  -- Executing Dial("IAX2/firefly at 89280250/3", "SIP/800|60") in new stack
Dec 22 17:22:10 NOTICE[6568]: app_dial.c:800 dial_exec: Unable to create 
channel of type 'SIP' (cause 3)
   == Everyone is busy/congested at this time

I have X-lite running and can call from one to the another sip so thats 
not a problem :(

> 
> also.. why disallow all and then allow most everything? seems like you 
> are trying to over think things.. no offense.

None taken. The only reason I am doing it that way and I know it is not 
great for a productions system is that I am using a few different SIP 
phones both soft and hard to do testing.

> 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
>>
>> _______________________________________________
>> Asterisk-Users mailing list
>> Asterisk-Users at lists.digium.com
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>> To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users