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List Info
Thread: Re: SIP error codes interpreted by Asterisk as?
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| Re: SIP error codes interpreted by
Asterisk as? |
  United States |
2008-03-20 08:14:40 |
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.
Thank you,
Mark.
-----Original Message-----
From: asterisk-dev-bounces lists.digium.com
[mailto:asterisk-dev-bounces 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 lists.digium.com
> [mailto:asterisk-dev-bounces 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 edvina.net
> Edvina AB * Asterisk training * http://edvina.net
> Asterisk SIP masterclass * Orlando Florida * April
2008
>
>
> _______________________________________________
> --Bandwidth and Colocation Provided by http://www.api-digital.c
om--
>
> asterisk-dev mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-dev
>
>
> _______________________________________________
> --Bandwidth and Colocation Provided by http://www.api-digital.c
om--
>
> asterisk-dev mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-dev
---
* Olle E Johansson - oej edvina.net
* Cell phone +46 70 593 68 51, Office +46 8 96 40 20,
Sweden
_______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.c
om--
asterisk-dev mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-dev
_______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.c
om--
asterisk-dev mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-dev
|
|
| Re: SIP error codes interpreted by
Asterisk as? |
  Sweden |
2008-03-20 08:37:14 |
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 lists.digium.com
> [mailto:asterisk-dev-bounces 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 lists.digium.com
>> [mailto:asterisk-dev-bounces 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 edvina.net
>> Edvina AB * Asterisk training * http://edvina.net
>> Asterisk SIP masterclass * Orlando Florida * April
2008
>>
>>
>> _______________________________________________
>> --Bandwidth and Colocation Provided by http://www.api-digital.c
om--
>>
>> asterisk-dev mailing list
>> To UNSUBSCRIBE or update options visit:
>> http://lists.digium.com/mailman/listinfo/asterisk-dev
>>
>>
>> _______________________________________________
>> --Bandwidth and Colocation Provided by http://www.api-digital.c
om--
>>
>> asterisk-dev mailing list
>> To UNSUBSCRIBE or update options visit:
>> http://lists.digium.com/mailman/listinfo/asterisk-dev
>
> ---
> * Olle E Johansson - oej edvina.net
> * Cell phone +46 70 593 68 51, Office +46 8 96 40 20,
Sweden
>
>
>
>
> _______________________________________________
> --Bandwidth and Colocation Provided by http://www.api-digital.c
om--
>
> asterisk-dev mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-dev
>
>
> _______________________________________________
> --Bandwidth and Colocation Provided by http://www.api-digital.c
om--
>
> asterisk-dev mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-dev
---
* Olle E Johansson - oej edvina.net
* Cell phone +46 70 593 68 51, Office +46 8 96 40 20,
Sweden
_______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.c
om--
asterisk-dev mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-dev
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