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

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m1278468 at allmail.net
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Posted: Mon May 02, 2016 10:53 pm Post subject: [asterisk-users] Migrating asterisk 11 to 13: some callers g

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 -- _____________________________________________________________________ -- 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

jcolp at digium.com
Guest

Posted: Tue May 03, 2016 6:10 am Post subject: [asterisk-users] Migrating asterisk 11 to 13: some callers g

Michael Maier wrote:

<snip>

Quote:

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

--
_____________________________________________________________________
-- 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

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m1278468 at allmail.net
Guest

Posted: Tue May 03, 2016 1:48 pm Post subject: [asterisk-users] Migrating asterisk 11 to 13: some callers g

On 05/03/2016 at 05:43 PM Joshua Colp wrote:

Quote:

Michael Maier wrote:

Quote:

On 05/03/2016 at 04:50 PM Joshua Colp wrote:

Quote:

Michael Maier wrote:

Quote:

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.

Quote:

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 Smile

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

--
_____________________________________________________________________
-- 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
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jcolp at digium.com
Guest

Posted: Tue May 03, 2016 1:52 pm Post subject: [asterisk-users] Migrating asterisk 11 to 13: some callers g

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

Michael Maier wrote:

<snip>

Quote:

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.

Quote:

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.

Quote:

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.

Quote:

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.

Quote:

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

--
_____________________________________________________________________
-- 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

ewieling at nyigc.com
Guest

Posted: Tue May 03, 2016 2:07 pm Post subject: [asterisk-users] Migrating asterisk 11 to 13: some callers g

I don't know the default setting for progressinband in the code, but it
is documented in Asterisk 11's sip.conf.sample as defaulting to never.
Maybe the docs were fixed since Asterisk 11.

from 11.21.x sip.conf.sample:

;progressinband=never ; If we should generate in-band ringing
always
; use 'never' to never use in-band
signalling, even in cases
; where some buggy devices might not
render it
; Valid values: yes, no, never Default:
never

On 05/03/2016 02:52 PM, Joshua Colp wrote:

Quote:

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

Michael Maier wrote:

<snip>

Quote:

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

Quote:

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.

Quote:

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.

Quote:

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.

Quote:

If I remove the option progrssinband=never via FreePBX, I can't find

any

Quote:

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.

Quote:

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.

--
if at first you don't succeed, skydiving isn't for you

--
_____________________________________________________________________
-- 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

jcolp at digium.com
Guest

Posted: Tue May 03, 2016 2:16 pm Post subject: [asterisk-users] Migrating asterisk 11 to 13: some callers g

Eric Wieling wrote:

Quote:

I don't know the default setting for progressinband in the code, but it
is documented in Asterisk 11's sip.conf.sample as defaulting to never.
Maybe the docs were fixed since Asterisk 11.

The behavior change to actually do what the option was documented to do.
As part of that the default was changed to reflect the past behavior,
thus why it was changed to no. The commit itself:

chan_sip: make progressinband default to no

After the "progressinband" value setting of "never" was updated to never
send a 183 this separated its use from the "no" value. Since "never" was
the default, but most users probably expect "no" this patch updates the
default for the "progressinband" setting to "no."

This was tracked under ASTERISK-24835[1].

[1] https://issues.asterisk.org/jira/browse/ASTERISK-24835

--
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

--
_____________________________________________________________________
-- 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

m1278468 at allmail.net
Guest

Posted: Wed May 04, 2016 2:12 am Post subject: [asterisk-users] Migrating asterisk 11 to 13: some callers g

On 05/03/2016 at 09:16 PM Joshua Colp wrote:

Quote:

Eric Wieling wrote:

Quote:

I don't know the default setting for progressinband in the code, but it
is documented in Asterisk 11's sip.conf.sample as defaulting to never.
Maybe the docs were fixed since Asterisk 11.

But "never" option therefore sends 180 Ringing which I was missing. The
new default "no" doesn't send 180 Ringing any more ... .

Quote:

Since "never" was
the default, but most users probably expect "no" this patch updates the
default for the "progressinband" setting to "no."

This was tracked under ASTERISK-24835[1].

[1] https://issues.asterisk.org/jira/browse/ASTERISK-24835

This makes sense! I migrated from

asterisk11-11.8.1-40_centos6.x86_64,

which had the default progressinband=never to

asterisk13-core-13.7.2-1.shmz65.1.94.x86_64

which had the new default.

POTS callers advertise support for early media - mobile callers on the
other hand don't advertise it, therefore mobile wasn't a problem because
early media (183) isn't triggered (and used!) at all.

Two strange things being left:

1. Why does progressinband=no work, if there is *no* ringgroup between
trunk and extension. This seems to be a "feature" of FreePBX.

2. Why is early media used even if the caller doesn't advertise it? Are
there other triggers like P-Early-Media?

Another basic question:
What do I need early media exactly for? I'm only using SIP phones -
nothing else. Couldn't it be completely disabled for these trunks? Or
would it break things like voice mail service e.g.? How can I disable it
completely even if it is advertised by the caller?

Thanks,
Michael

--
_____________________________________________________________________
-- 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

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