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

[David Uzzell]

Sorry for replying to my own mesg but I have more info.

David Uzzell wrote:
> Matt Hess wrote:
> 
>> is this current cvs or something? It looks completely abnormal for 
>> stable..
>>
> 
> Ah sorry it is CVS 12/12/04
> 

I have just this min downloaded the latest CVS as of about 20mins ago 
and compiled etc and the error is now gone.

I am unsure what the error was but it must have been something to do 
with that CVS version.

Thanks, now I just have to figure out the default dialplan :)

David



>> 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
>>>
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