[asterisk-dev] 1.6.1.20 regression: Tones not always generated

[Pavel Troller]

> On Tue, May 25, 2010 at 12:20 PM, Pavel Troller <patrol at sinus.cz> wrote:
> > After reverting just this single change, i.e. moving the ast_prod() call,
> > where it was, all the tones are on their correct places again. I'm aware of
> > that I've reintroduced the local channel deadlock, but it is less important
> > for me (not using chan_local very often) than missing tones.
> >
> Do you see the same problem is you use the 1.6.2 branch?  I only ask
> because 1.6.0 and 1.6.1 are now closed for bug fixes (only security).
> If this is a regression, the fix would be applied to 1.6.2.
> 
> I would also create a new ticket on https://issues.asterisk.org to
> track any potential fixes.

Hi Paul,
  the problem is that I currently don't operate any 1.6.2.x system. I have
about 12 1.6.1.x installations, and my plan was to run them until 1.8 is out -
I don't see so much new things in 1.6.2 to convince me for upgrading, 1.6.1 is
otherwise stable in my environment and I'm happy with it. Upgrading asterisk
is not very easy for me, I have many patches in it, which I would have to adapt
(for example, I have all the asterisk self-contained in /opt/asterisk without
any files dropped to the general system directories like /etc, /lib etc., which
still cannot be done with just configure --prefix=/opt/asterisk because of
paths hardcoded to the source, documentation etc.). 
  To help with searching for a possible cause, I've made two calls to the
Ringing() app, one with the bug present, and the other without it (with the
ast_prod() call moved back to its original place). The non-functional version
produces the following debug:
  
[May 25 18:17:00] DEBUG[31312] pbx.c: Launching 'Ringing'
[May 25 18:17:00] DEBUG[31312] channel.c: Driver for channel 'SIP/2122-00000004' does not support indication 3, emulating it
[May 25 18:17:00] DEBUG[31312] channel.c: Set channel SIP/2122-00000004 to write format slin
[May 25 18:17:00] DEBUG[31312] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second
[May 25 18:17:00] DEBUG[31312] channel.c: Prodding channel 'SIP/2122-00000004'
[May 25 18:17:00] DEBUG[31312] channel.c: Set channel SIP/2122-00000004 to write format alaw
[May 25 18:17:00] DEBUG[31312] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[May 25 18:17:00] DEBUG[31312] pbx.c: Launching 'Wait'
[May 25 18:17:01] DEBUG[31312] rtp.c: Got RTCP report of 52 bytes
... and nothing regarding RTP being sent

  The functional one looks like this:

[May 26 06:25:45] DEBUG[28291] pbx.c: Launching 'Ringing'
[May 26 06:25:45] DEBUG[28291] channel.c: Driver for channel 'SIP/2122-00000018' does not support indication 3, emulating it
[May 26 06:25:45] DEBUG[28291] channel.c: Prodding channel 'SIP/2122-00000018'
[May 26 06:25:45] DEBUG[28291] channel.c: Set channel SIP/2122-00000018 to write format slin
[May 26 06:25:45] DEBUG[28291] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second
[May 26 06:25:45] DEBUG[28291] pbx.c: Launching 'Wait'
[May 26 06:25:45] DEBUG[28291] rtp.c: Ooh, format changed from unknown to alaw
[May 26 06:25:45] DEBUG[28291] rtp.c: Created smoother: format: 8 ms: 20 len: 160
[May 26 06:25:45] DEBUG[28291] rtp.c: Got RTCP report of 52 bytes
... and RTP is happily running.

These debugs were created by calling this extension:
exten => 13006,1,Macro(stdserv,0,"Ring Tone Check")
exten => 13006,n,Proceeding()
exten => 13006,n,Wait(1)
exten => 13006,n,Progress()
exten => 13006,n,Wait(1)
exten => 13006,n,Ringing()
exten => 13006,n,Wait(60)
exten => 13006,n,Hangup()

(you can ignore the macro, it is not important for our case).

Do you think I have really create the ticket, even I didn't verify the 1.6.2
version (yet) ?

With regards, Pavel