[asterisk-dev] pseudo channel

[john at spectross.com]

Hi Steve,

Thanks for the reply.

What I don't understand is:
1) Why are commands (ioctls) like "tone_zone_play_tone" called over pseudo 
channels (refer zt_call: RBT for CALL_WAIT) ?

/* 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);

2) How can we obtain events from non-existent pseudo channels ?

/*zt_handle_event*/
res = zt_get_event(p->subs[index].zfd);

Regds.
John

----- Original Message ----- 
From: "Steve Davies" <davies147 at gmail.com>
To: <john at spectross.com>; "Asterisk Developers Mailing List" 
<asterisk-dev at lists.digium.com>
Sent: Tuesday, March 17, 2009 7:58 PM
Subject: Re: [asterisk-dev] pseudo channel


2009/3/17  <john at spectross.com>:
> Hi,
>
> I was trying to understand the chan_zap/chan_dahdi code.
>
> What is the point of sending commands / receiving events / reading & 
> writing
> voice packets over pseudo ("/dev/zap/pseudo") channels ?
>

Person A: What is the most important thing a comedian needs to know?

Person B: I don't know what is the most....

Person A (interrupts): TIMING!


I believe that /dev/zap/pseudo provides access to timing for Asterisk
components that do not have a reliable clock source, without requiring
or using up an actual voice resource from the zap/dahdi hardware. This
is useful (for example) for Meetme where you can have attendees
arriving and leaving, so cannot rely on anyone as a clock source.

(I hope that is right)

Regards,
Steve