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

[Tzafrir Cohen]

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

-- 
               Tzafrir Cohen
icq#16849755              jabber:tzafrir.cohen at xorcom.com
+972-50-7952406           mailto:tzafrir.cohen at xorcom.com
http://www.xorcom.com  iax:guest at local.xorcom.com/tzafrir