[asterisk-dev] SIP error codes interpreted by Asterisk as?

[Johansson Olle E]

20 mar 2008 kl. 13.34 skrev Mark Hamilton:

> Hi Johansson,
>
> I just replied to Antonio thinking ISDN is not related to SIP error  
> codes,
> but your message says that you do infact use the SIP error codes.
>
> How reliable is this information provided by the ISDN hangup causes?

I am at loss on how to answer, guess I don't really understand the  
question.
We do translate from SIP2ISDN and from ISDN2SIP all the time. When  
Asterisk
in itself the cause of the hangup or the disconnect, we've tried to  
make it
as good as possible, but there's propably room for additions and  
changes.
Feedback on the implementation is more than welcome, so that we can add
proper hangupcauses and error codes in more parts of Asterisk.

/Olle

>
>
> Thank you,
> Mark.
>
> -----Original Message-----
> From: asterisk-dev-bounces at lists.digium.com
> [mailto:asterisk-dev-bounces at lists.digium.com] On Behalf Of  
> Johansson Olle E
> Sent: March 20, 2008 3:15 AM
> To: Asterisk Developers Mailing List
> Subject: Re: [asterisk-dev] SIP error codes interpreted by Asterisk  
> as?
>
>
> 19 mar 2008 kl. 23.50 skrev Mark Hamilton:
>
>> Hello,
>>
>> There are quite a few error codes returned upon calling a number.
>> Like SIP 404 Not Found, 486 Temporarily Unavailable, etc. How does
>> Asterisk interpret these SIP error codes?
>>
>> Because when a call is made through Asterisk, and let's say it's a
>> disconnected number, Asterisk returns "Call failed, reason 0" or a
>> reason number from 1-10. What makes Asterisk decide what reason is a
>> SIP error code classified as?
>
> In chan_sip we translate those error codes to the ISDN hangup causes
> we use internally in Asterisk, which you can reach
> in the HANGUPCAUSE dialplan variable. We follow the IETF standards
> where available, or the Cisco implementation.
>
> On the other end of the call, the ISDN cause codes are translated back
> to SIP.
>
> /Olle
>
>
> -------
> oej at edvina.net
> Edvina AB * Asterisk training * http://edvina.net
> Asterisk SIP masterclass * Orlando Florida * April 2008
>
>
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---
* Olle E Johansson - oej at edvina.net
* Cell phone +46 70 593 68 51, Office +46 8 96 40 20, Sweden