asterisk users - Feb 2016 - How to deal with error messages passed as Early Media

Hello,

I'm trunking with an ITSP that, when treating an outbound to an unknown
destination, either:
- send a SIP error code (I can't be more explicit, at the moment),
- or cast a pre-recorded audio message using Early Media.

At the same time, I'm also trunking with Contact Center solution which
doesn't support Early Media.


Beside asking my ITSP to treat calls consistently or ask  Contact Centerto
support Early Media, is there a way to configure Asterisk to unify both
above error treaments into a single one ?

How can I best deal with error messages passed as Early Media.

Best regards
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On 03/02/16 14:41, Olivier wrote:
> How can I best deal with error messages passed as Early Media.
Tell the ITSP to give you proper signaling, if they wont then get a new 
ITSP. I suspect if they can't handle this correctly, there will be a lot 
more they're doing wrong as well. Long term you'll save yourself a whole
lot of bother.

Steve

2016-02-03 15:59 GMT+01:00 Steve Howes <steve-lists at geekinter.net>:
>
>
> On 03/02/16 14:41, Olivier wrote:
>
>> How can I best deal with error messages passed as Early Media.
>>
> Tell the ITSP to give you proper signaling, if they wont then get a new
> ITSP. I suspect if they can't handle this correctly, there will be a
lot
> more they're doing wrong as well. Long term you'll save yourself a
whole
> lot of bother.
>
>
Yes but I'm afraid that, in this industry, the rule is to pass anything
received to the other party.


Steve
>
> --
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For calls that fail, even where early media is played, the call should
terminate with a 4xx or 5xx SIP response which to a certain degree
correlates to the nature of the actual failure.  The SIP error code is
delayed until the media playback completes, but should be no different
whether or not early media is used (for the same actual failure).

Early media is simply an audio stream for human consumption to explain the
failure.  There should be no need to attempt to recognize it, unless your
ITSP is not terminating the call correctly.



On Wed, Feb 3, 2016 at 8:41 AM, Olivier <oza.4h07 at gmail.com> wrote:
> Hello,
>
> I'm trunking with an ITSP that, when treating an outbound to an unknown
> destination, either:
> - send a SIP error code (I can't be more explicit, at the moment),
> - or cast a pre-recorded audio message using Early Media.
>
> At the same time, I'm also trunking with Contact Center solution which
> doesn't support Early Media.
>
>
> Beside asking my ITSP to treat calls consistently or ask  Contact Centerto
> support Early Media, is there a way to configure Asterisk to unify both
> above error treaments into a single one ?
>
> How can I best deal with error messages passed as Early Media.
>
> Best regards
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>                http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>


-- 
[image: Digium logo]
Scott Griepentrog
Digium, Inc ? Software Developer
445 Jan Davis Drive NW ? Huntsville, AL 35806 ? US
direct/fax: +1 256 428 6239 ? mobile: +1 256 580 6090
Check us out at: http://digium.com ? http://asterisk.org
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On 04/02/16 06:00, Scott Griepentrog wrote:
> For calls that fail, even where early media is played, the call should 
> terminate with a 4xx or 5xx SIP response which to a certain degree 
> correlates to the nature of the actual failure.  The SIP error code is 
> delayed until the media playback completes, but should be no different 
> whether or not early media is used (for the same actual failure).
>
> Early media is simply an audio stream for human consumption to explain 
> the failure.  There should be no need to attempt to recognize it, 
> unless your ITSP is not terminating the call correctly.
>
I recently ran some analysis of early media messages found in cellular 
networks by recording the calls and running it through CMU-Sphinx. There 
are only a few types of early media messages per network, to cover a 
raft of failures.

But I always got a SIP error code back as well, the early media I found 
tended to play for upto 20 secs then drop the call, then you get the 
error code. It might not be a very descriptive error but its still an 
error code, and the early media audio message is not always very 
distinct either. In the GSM networks the GSM failure code is more useful 
but still seems somewhat randomly assigned by the provider, even 
including the odd temporary failures.

You are not really worried about what the failure reason is, its the 
caller who needs to decide - did I misdial, is the number really 
disconnected, is their phone out of coverage etc, you just need to try 
your next available network, if the caller hasn't already hung up after 
hearing the message

You could possibly examine the audio before you get an answer but then 
you might get caught by some other system or PBX playing early media 
before answer that isn't actually a failure.

If your ITSP is not giving you an error code then you have an issue.

Cheers Duncan
>
>
> On Wed, Feb 3, 2016 at 8:41 AM, Olivier <oza.4h07 at gmail.com 
> <mailto:oza.4h07 at gmail.com>> wrote:
>
>     Hello,
>
>     I'm trunking with an ITSP that, when treating an outbound to an
>     unknown destination, either:
>     - send a SIP error code (I can't be more explicit, at the moment),
>     - or cast a pre-recorded audio message using Early Media.
>
>     At the same time, I'm also trunking with Contact Center solution
>     which doesn't support Early Media.
>
>
>     Beside asking my ITSP to treat calls consistently or ask  Contact
>     Centerto support Early Media, is there a way to configure Asterisk
>     to unify both above error treaments into a single one ?
>
>     How can I best deal with error messages passed as Early Media.
>
>     Best regards
>
>     --
>     _____________________________________________________________________
>     -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>     New to Asterisk? Join us for a live introductory webinar every Thurs:
>     http://www.asterisk.org/hello
>
>     asterisk-users mailing list
>     To UNSUBSCRIBE or update options visit:
>     http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
>
> -- 
> Digium logo
> Scott Griepentrog
> Digium, Inc ? Software Developer
> 445 Jan Davis Drive NW ? Huntsville, AL 35806 ? US
> direct/fax: +1 256 428 6239 ? mobile: +1 256 580 6090
> Check us out at: http://digium.com ? http://asterisk.org
>
>
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