[asterisk-users] Migrating asterisk 11 to 13: some callers get no ringback tone any more
ewieling at nyigc.com
Tue May 3 14:07:09 CDT 2016
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
; use 'never' to never use in-band
signalling, even in cases
; where some buggy devices might not
; Valid values: yes, no, never Default:
On 05/03/2016 02:52 PM, Joshua Colp wrote:
> Whoops, email client auto-filled dev previously. Let's try this again.
> Michael Maier wrote:
> > 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
> > 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
> > 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.
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