asterisk users - Aug 2004 - <<< MEETME_AGI_BACKGROUND inside MEET ME>>>

Howdie:

I've been reading some old threads and still have a couple of questions 
about applying the AGI_BACKGROUND script inside a Conference. Perhaps 
someone can save me a bit of fidd'lin.

Am I right in assuming that the MEETME_AGI_BACKGROUND script **WILL WORK** 
on SIP conferenced channels **WITHOUT** an **ACTIVE** zap channel-- AS LONG 
AS THERE IS A DIGIUM CARD INSTALLED IN THE ASTERISK BOX?

This seems to be the drift of the following thread:

http://www.mail-archive.com/asterisk-users@lists.digium.com/msg33930.html

Also--

Has anyone had any success implementing a WAIT FOR DIGIT AGI command 
solution **INSIDE** a conference via **PHP?** The solution (I Guess) would 
look something like the following thread:

http://www.mail-archive.com/asterisk-users@lists.digium.com/msg33772.html

And Finally--

If the above WAIT FOR DIGIT, MEETME_AGI_BACKGROUND does indeed work for 
someone-- will the AGI accept DTMF from any and **ALL** conference partys-- 
or are there restrictions to which connections can actually pass DTMF?

Hey Thanks-- Any insight on these issues would be ** GREATLY** appreciated.

Cheers-
JJQ

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In article <BAY15-F18yKDpsCMYRB00057045@hotmail.com>,
jeff quade <jjq90@hotmail.com> wrote:
> Howdie:
> 
> I've been reading some old threads and still have a couple of questions
> about applying the AGI_BACKGROUND script inside a Conference. Perhaps 
> someone can save me a bit of fidd'lin.
> 
> Am I right in assuming that the MEETME_AGI_BACKGROUND script **WILL WORK** 
> on SIP conferenced channels **WITHOUT** an **ACTIVE** zap channel-- AS LONG
> AS THERE IS A DIGIUM CARD INSTALLED IN THE ASTERISK BOX?
No, that's not correct.

The merging of incoming voice packets and the distribution of the merged
packets to the output channels is done within the Zaptel driver. For
true Zap channels, this happens completely at the driver layer. For a
non-Zap channel, app_meetme creates a Zap pseudo-channel for the driver
to use, and then copies the audio data in both directions between the
non-Zap channel and its associated pseudo channel. You can see this in
the middle of the conf_run() function.

The catch is that conf_run() EITHER calls the AGI script, OR runs the
pseudo-channel copying loop, but not both. So if you use an AGI script,
there is nothing linking the non-Zap channel to its associated pseudo.

It *is* necessary either to have a Digium card or a dummy timing driver
(e.g. ztdummy or zaprtc) in order for MeetMe to work at all, but that
doesn't help you use AGI with SIP channels.

Cheers
Tony
-- 
Tony Mountifield
Work: tony@softins.co.uk - http://www.softins.co.uk
Play: tony@mountifield.org - http://tony.mountifield.org

Tony/et-al--

Thanks for the response/info so far on the AGI_BACKGROUND issue.

If I read you correctly--

SIP channels **May Conference** by using the ztDummy driver 
(pseudo-driver)-- but by doing so, the conferenced channels have no capacity 
to pass DTMF into an associated AGI (Tony explained this nicely!)

ZAP channels **May Conference** and because of the driver design, **May Pass 
DTMF** via an associated _AGI_BACKGROUND script--

So I would guess, (please point me to docs, if any) that **EACH** true Zap 
channel within a conference may be associated with **ITS OWN** 
MEETME_AGI_BACKGROUND script-- which could monitor the DTMF on its 
individual channel--??--.. Is this correct?

The MEETME_AGI_BACKGROUND script is NOT conference oriented BUT connection 
(channel) oriented--??--??..

THE PRACTICAL APPLICATION here is to have a ZAP channel conference LEADER, 
who would be able to launch "canned" audio presentations (of regional
sales
data) into an existing conference, via a PHP AGI_BACKGROUND script using the 
"GET DATA" or "STREAM FILE" AGI commands.

If this sounds plausible, Im gonna fiddle with it today/week-end (Im on 
NY-Time)

Again-- Thanks for the insight.

Cheers-
JJQ

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Howdie-

Spent a couple of days on this issue:
>THE PRACTICAL APPLICATION here is to have a ZAP channel conference LEADER, 
>who would be able to launch "canned" audio presentations (of
regional sales
>data) into an existing conference, via a PHP AGI_BACKGROUND script using 
>the "GET DATA" or "STREAM FILE" AGI commands.
It seems (at least I cant find a way) that there is **NO WAY** to play 
pre-recorded audio into a conference **Through manipulation of the 
dialplan*** via AGI or directly via a new channel set-up from Manager.

As far as I can tell upon review of the source, suggestions from the group, 
and just plain trial and error:

The MeetMe MUX (ie: careful write() function in MeetMe) happily mixes SIP or 
ZAP channel voice traffic from within the main Channel thread-- but bypasses 
the **CONFRENCE MUX** when audio is sent via any MEETME_AG_BACKGROUND script 
**FORKED OFF** of the main channel thread.

(Upon review of the source this makes sense! As Tony suggested its **NOT** 
launching the AGI thats a problem-- Its the audio MUX.)

Seems to me there has got to be a simple way to get this functionality 
without hacking into the source---- HAve I missed something obvious? Perhaps 
something via a loopback local channel?

Any ideas?

Thanks--
JJQ

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Tony-

Your .call file solution via loopback works **JUST FINE!**
In fact GREAT. Thanks.

This solution is a bit twisted-off-of what I had in mind for control and 
launching of the audio files via the MeetMe Background Script, but may 
actually work **BETTER** for my needs. It will actually free-up the MASTER 
ZAP control channel to send and recieve AGI commands without the added 
burden (and limited functionality) of playing back the audio file on the 
same channel.

This solves the audio mixing issue (bypass of AGI Audio within MeetMe)-- but 
more importantly provides a nice way to key-up the canned audio via 
extensions in the dialplan-- **OR** perhaps a more sophisticated use of an 
AGI which sets a Channel Variable which maps to an audio file-- then 
executes the same extension as a sorta-file-launcher.

Ill have to sleep on it before I charge into the PHP.

Boy, it would be nice to have a simple PDF diagram of the data flow through 
asterisk and into MeetMe (ala the rtp_flow.pdf in the source.)

Thanks Again, hope your solution provides the group with new fodder for neat 
new apps.

Cheers-
JJQ

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