[asterisk-dev] pseudo channel

[john at spectross.com]

Hi Russell,

Thanks for the reply.
 __________                 ________________                     __________
|                    |                |                                | 
|                    |
|        A         |                |              B                | 
|        C         |
|  (Phy. Ch.)  |                |        (Phy. Ch.)         | 
|  (Phy. Ch.)  |
|__________|                |________________| 
|__________|
          |                               |                        | 
|
  SUB_REAL          SUB_REAL   SUB_CALLWAIT         SUB_REAL
          |                               |                        | 
|
     ___|___                  ___|___            ___|___ 
___|___
     |            |                 |            |           | 
|                   |            |
     |AC-R  |-------- -> |AC-R  |           |AC-CW |<-----------| AC-R  |
     |______|  app_dial  |______|           |_______|   app_dial   |______|

The blocks named A, B & C are used to represent physical channels. The 
blocks below represent
ast_channels (AC) corresponding to SUB_REAL (R) & SUB_CALLWAIT (CW) 
sub-channels.

Could you please help me understand how could we generate a ring back tone 
for 'C' by using an 'ioctl'
over the 'pseudo file decriptor' corresponding to 'SUB_CALLWAIT' of 'B'.

Well it could be possible after bridging but this case is pre-bridge i.e. 
during ast_call (zt_call).

I also tried commenting this piece of code but saw no difference in the 
behaviour of call-wait (i.e. RBT was always present).

After further investigation it seems that app_dial is the one responsible 
for RBT generation to caller in all cases.

Plz. clarify.

Kindly also throw some light over reception of events over pseudo 
file-descriptors corresponding to sub-channels.

Regds.
John

----- Original Message ----- 
From: "Russell Bryant" <russell at digium.com>
To: <john at spectross.com>; "Asterisk Developers Mailing List" 
<asterisk-dev at lists.digium.com>
Sent: Tuesday, March 17, 2009 10:21 PM
Subject: Re: [asterisk-dev] pseudo channel


> john at spectross.com wrote:
>> /* Make ring-back */
>> if (tone_zone_play_tone(p->subs[SUB_CALLWAIT].zfd, ZT_TONE_RINGTONE))
>>  ast_log(LOG_WARNING, "Unable to generate call-wait ring-back on channel 
>> %s\n", ast->name);
>
> If you read my previous reply, this piece of code will start to make 
> sense.
>
> In this case, we're dealing with call waiting.  That means that there is 
> an Asterisk channel already up and connected to SUB_REAL, which is the 
> DAHDI channel associated with the physical interface.  Now, we have 
> another call coming in destined for this interface.  We're going to go 
> ahead and set up this call with a psuedo channel.  The piece of code you 
> refer to tells DAHDI to send ring back to the caller.  Meanwhile, the 
> DAHDI channel associated with the physical interface is hearing a call 
> waiting beep.
>
> If this still isn't clear, I can try to draw up some ASCII art that 
> demonstrates the audio flow here ...
>
> -- 
> Russell Bryant
> Digium, Inc. | Senior Software Engineer, Open Source Team Lead
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> Check us out at: www.digium.com & www.asterisk.org