[asterisk-dev] Listening to a remote dialtone on a DAHDI/PRI has been suppressed in 1.6.1 ?

[Pavel Troller]

> On Wed, May 06, 2009 at 06:02:58AM +0200, Pavel Troller wrote:
> > Hi!
> >   I'm new on this list, let me introduce myself shortly first.
> >   I'm a switching systems expert, working for one of major operators in Czech
> > Republic as well being an associated professor at Czech Technical University,
> > lecturing the same subject. My main focus is concentrated on "big" switches,
> > like Siemens EWSD or Nortel CallServer 2000, as well as some PBXes, especially
> > Ericsson MD110/MX1, Siemens HiPath etc. I'm working with Asterisk mainly as my
> > hobby, but I'm also using it in my work, currently we run a Meetme conference
> > and Customer Announcement System on Asterisk at our telco.
> >   And now, my first question: I've upgraded my Asterisk from 1.6.0.9 to 1.6.1.0
> > and I've found that I cannot listen to the dialtones from remote exchanges on 
> > overlapped outgoing calls over DAHDI/PRI (i.e. Dial(DAHDI/g1) without a number,
> > with empty SETUP sent). 
> >   I've reviewed chan_dahdi.c and I've found, that the following change:
> > 
> >         case SIG_PRI:
> >         case SIG_BRI:
> >         case SIG_BRI_PTMP:
> >         case SIG_SS7:
> >                 /* We'll get it in a moment -- but use dialdest to store pre-setup_ack digits */
> >                 p->dialdest[0] = '\0';
> > +               p->dialing = 1;
> >                 break;
> > 
> > around line 2710 is causing this. I've commented this line out and I have my
> > dialtones back.
> >   I would like to ask, why this change has been made and whether it is 
> > necessary. I'm using overlap mode dialling very often and I (and my users too)
> > like to hear the dialtones as we are traversing through our networks :-). Is
> > there another way to open the audio path immediately, i.e. using some new 
> > config option, yet unknown to me ? 
> 
> 'When was this changed'? svn blame to the rescue.
> 
>   svn blame http://svn.digium.com/svn/asterisk/branches/1.6.1/channels/chan_dahdi.c
>   
> Shows, among others:
> 
>   183333   tilghman               p->dialing = 1;
> 
> 
>   svn log -r 183333 http://svn.digium.com/svn/asterisk/branches/1.6.1/
> 
>   Delay signalling progress until a PRI channel really signals progress.
>   (closes issue #13034)
>   Reported by: klaus3000
>   Patches:
>     20090316__bug13034.diff.txt uploaded by tilghman (license 14)
>     patch_trunk_183progress_klaus3000.txt uploaded by klaus3000 (license 65)
>   Tested by: klaus3000
> 
Hi!
  Hmm, I see. It was a bug fix.
  But from my point of view, it introduced a new bug. It's interesting
that I never encountered a problem with #13034, I've always heard all the tones
what I expected to hear, but now I don't.
  So, what next ? Is it possible to fix my issue without reintroducing the
former bug ? Should I file a new bug ?
  With regards, Pavel Troller