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

[Johansson Olle E]

20 mar 2008 kl. 14.14 skrev Mark Hamilton:

> Hi Olle,
>
> The thing is, we use VoIP completely. All our systems are over IP,  
> and so we
> have no T1, PRI, etc.
> So, accordingly, quite a few times when a call fails, Asterisk seems  
> to
> classify it under 'reason 0'. Be it congestion, or disconnected  
> number, or
> sometimes even a fast busy gets classified as reason 0.
>
> I'm trying to remove disconnected numbers from our contact lists so  
> we can
> actually get by through email or something. Either way, I really  
> want to
> somehow figure out what the disconnected numbers are. And with  
> almost all
> failed calls getting in the 'reason 0' category, it's hard.
>
> Any suggestions would be greatly appreciated.

Mark,
Debug it down and see what error code you get from the carrier.
Then post a bug report. If there's an error HANGUPCAUSE should
catch it.

Happy Easter!
/O
>
>
> 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 8:51 AM
> To: Asterisk Developers Mailing List
> Subject: Re: [asterisk-dev] SIP error codes interpreted by Asterisk  
> as?
>
>
> 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
>
>
>
>
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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