[asterisk-dev] [asterisk-users] Migrating asterisk 11 to 13: some callers get no ringback tone any more

[James Finstrom]

Update to the latest core/ringgroups version (may be in edge still)  This
will correct the ringback issues.  This ultimately comes down to how your
provider handles these messages. Some (not all) providers freak out if they
get "progress" and "ringing".  Recently we added progress in ringgroups. If
you have ringing on your inbound route you are now sending both and this
makes some providers cry.  So the latest iteration does NOT send progress
if you set ringing to be true. Ultimately this is a provider problem as a
direct result of configuration in FreePBX and not an Asterisk issue.

On Tue, May 3, 2016 at 9:47 AM, Joshua Colp <jcolp at digium.com> wrote:

> Michael Maier wrote:
>
> <snip>
>
> Ok - but this doesn't seem to answer my main question:
>>
>> Why must
>>
>> progressinband=never
>>
>> be applied especially if asterisk uses it by default? The big difference
>> between w/ and w/o it is:
>>
>
> The default in 13 is "no" which still allows early media through. That
> option has a complicated past.
>
>
>> w/o the option progrssinband=never just the sip-package
>>         183 Session Progress
>> is sent.
>>
>
> Yes, because it's doing inband progress using a media stream.
>
>
>> w/ the option set, the additional sip-packages
>>         100 Trying
>>         180 Ringing
>>         180 Ringing
>> are sent.
>>
>> If progrssinband=never is the default, the Ringing package should be
>> sent always, shouldn't it?
>>
>
> It's not the default.
>
>
>> If I remove the option progrssinband=never via FreePBX, I can't find any
>> other value provided to progrssinband in /etc/asterisk/*.
>>
>>
>> Why does it work always correctly w/ the second trunk, which is
>> connected directly to the extension?
>>
>
> FreePBX may not use inband progress for that scenario, causing it to send
> out of band ringing instead.
>
>
>> Is it possible to switch off the standard behavior of asterisk /
>> progrssinband for ring groups only by setting some other options?
>>
>
> Asterisk does not have the concept of ring groups, this is a FreePBX
> construc. Asterisk itself does allow the option to be set on an individual
> basis for the entries in sip.conf.
>
>
> --
> Joshua Colp
> Digium, Inc. | Senior Software Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
> Check us out at: www.digium.com & www.asterisk.org
>
>
> --
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-- 
James
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