asterisk users - May 2016 - Migrating asterisk 11 to 13: some callers get no ringback tone any more

Hello!


I migrated asterisk 11 to 13 as user of FreePBX 12.0.76.2.

As customer of German Telekom, I have three numbers and therefore three
trunks - each number is bound to one trunk.

After migration, some callers complained about missing ringback tone:
they didn't hear any ring tone and where surprised that they suddenly
got me anyway :-). The connection afterwards was as expected.


Deeper investigation yielded, that a few caller groups have been
affected by this problem:

- All POTS customers of German Telekom
- VoIP customer of T-Systems (usually companies which transfered their
  telephone system)

Not affected have been callers like All-IP customers of German Telekom
or any tested mobile caller or caller using other telecommunications
companies.


To make it even more strange: Calls coming from T-Systems customer via
call forwarding have been working fine, too.

And:
The ringback tone wasn't missing, if the second number (the second
trunk) of the asterisk installation was used!

The only difference between those two trunks is: The first trunk is
configured to a ring group - the second trunk is configured directly to
an extension.



My solution after long time of digging around:
I added progressinband=never to sip_general_additional.conf

But this solution confuses me, because

http://www.voip-info.org/wiki/view/Asterisk+sip+progressinband

tells:

progressinband=never

Whenever ringing occurs, send "180 ringing" as long as "200
OK" has not
yet been sent. This is the default behaviour of Asterisk.
			   ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^


Why do I have to provide it especially if it is the default behavior?
Why did it work without this option with asterisk 11? Why is there
suddenly a difference in behavior between binding a trunk to a ring
group or an extension?



Puzzled,
regards,
Michael Maier

Michael Maier wrote:

<snip>
>
> And:
> The ringback tone wasn't missing, if the second number (the second
> trunk) of the asterisk installation was used!
>
> The only difference between those two trunks is: The first trunk is
> configured to a ring group - the second trunk is configured directly to
> an extension.
>
>
>
> My solution after long time of digging around:
> I added progressinband=never to sip_general_additional.conf
>
> But this solution confuses me, because
>
> http://www.voip-info.org/wiki/view/Asterisk+sip+progressinband
>
> tells:
>
> progressinband=never
>
> Whenever ringing occurs, send "180 ringing" as long as "200
OK" has not
> yet been sent. This is the default behaviour of Asterisk.
> 			   ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
>
>
> Why do I have to provide it especially if it is the default behavior?
> Why did it work without this option with asterisk 11? Why is there
> suddenly a difference in behavior between binding a trunk to a ring
> group or an extension?
I'm not really sure what would be different, as that would be a FreePBX 
construct and not of Asterisk itself. If you provide a SIP debug of the 
non-working case I can see if anything is out of the ordinary in the 
signaling.

-- 
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

On 05/03/2016 at 05:43 PM Joshua Colp wrote:
> Michael Maier wrote:
>> On 05/03/2016 at 04:50 PM Joshua Colp wrote:
>>> Michael Maier wrote:
>>>> Hello Joshua!
>>>>
>>>>
>>>> I attached the sip debug without the progressinband=never set.
The
>>>> caller didn't get a ring back tone as expected.
>>> Please keep this on list so that anyone who may run into a similar
>>> problem in the future has a chance of finding this discussion.
>>
>> You are right - normally I'm going exactly this way. But I
don't want
>> the traces to be world wide readable (->  privacy). I will write a
>> summary to the list as far as we know more.
>>
>>> As for your log there's nothing of note really, it's just
expecting to
>>> send the ringing as inband audio instead of out of band. Does
"rtp set
>>> debug on" show the RTP traffic going to the other side?
>>
>> Yes. I attached it.
>>
>> And no - there isn't any packet blocked by iptables :-).
> 
> There is nothing abnormal here and Asterisk appears to be doing the
> correct thing. It's sending an audio stream with early progress to the
> caller. It may be that in a previous FreePBX, or when used with 13, they
> changed the behavior for this to force early media and the provider is
> not allowing it.
> 
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:

w/o the option progrssinband=never just the sip-package
	183 Session Progress
is sent.

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?

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?

Is it possible to switch off the standard behavior of asterisk /
progrssinband for ring groups only by setting some other options?



Thanks,
kind regards,
Michael

Whoops, email client auto-filled dev previously. Let's try this again.

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
construct. 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