<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
<html>
<head>
<meta content="text/html;charset=ISO-8859-1" http-equiv="Content-Type">
</head>
<body bgcolor="#ffffff" text="#000000">
Mark, Russel <br>
<br>
Thanks for the information.<br>
<br>
So now i'm looking into app_confbridge and another couple questions:<br>
<br>
1) On lines 276, 510 there are calls to play_sound_fiile...<br>
It seems to me they shoubd be conditioned on OPTION_QUIET<br>
<br>
2) Is there app_page equivalent using app_confbride,<br>
and if not what would be better approach, modify app_page to<br>
use a chan variable or an option argument to use confbridge in
place of meetme, or write a completley new version of app_page?<br>
<br>
Thanks<br>
Vadim<br>
<br>
<br>
<br>
Mark Michelson wrote:
<blockquote cite="mid:49B70EF1.2080102@digium.com" type="cite">
<pre wrap="">Vadim Lebedev wrote:
</pre>
<blockquote type="cite">
<pre wrap="">Hello,
I've just built asterisk form svn trunk on mac os/x 10.5 and have 3
problems:
1) when doing sudo asterisk --- it does not go into background but
stays attached to the terminal.
</pre>
</blockquote>
<pre wrap=""><!---->
I'm not familiar with this error...
</pre>
<blockquote type="cite">
<pre wrap="">2) when i do sudo asterisk -r , the other instance of atserisk says:
/-- Remote UNIX connection/
/
/
/ / However the isntances are not communicating between them.
Whatever i type on the console is ignored
/
/
/
/
</pre>
</blockquote>
<pre wrap=""><!---->
This is a known issue with Mac OS X, and seems to be a bug in their multiplexed
I/O support. Here is a bug report about it:
<a class="moz-txt-link-freetext" href="http://bugs.digium.com/view.php?id=13404">http://bugs.digium.com/view.php?id=13404</a>
All signs seem to point to the bug existing in Darwin and not in Asterisk since
the exact same code works on other platforms just fine and none of us have been
able to find anything in our code that is blatantly wrong.
</pre>
<blockquote type="cite">
<pre wrap="">3) MeetMe is not built -- menuselect says it depends on dahdi, but i
thought with new bridging and timing support it can work
without ztdummy or it's dahdi equivalent.
</pre>
</blockquote>
<pre wrap=""><!---->
You are incorrect. In trunk, there is a new conference bridging method that does
not require DAHDI. However, it is not app_meetme, it is app_confbridge.
app_meetme still requires DAHDI for audio mixing.
Mark Michelson
</pre>
<blockquote type="cite">
<pre wrap="">
Thanks
Vadim
------------------------------------------------------------------------
_______________________________________________
--Bandwidth and Colocation Provided by <a class="moz-txt-link-freetext" href="http://www.api-digital.com">http://www.api-digital.com</a>--
asterisk-dev mailing list
To UNSUBSCRIBE or update options visit:
<a class="moz-txt-link-freetext" href="http://lists.digium.com/mailman/listinfo/asterisk-dev">http://lists.digium.com/mailman/listinfo/asterisk-dev</a>
</pre>
</blockquote>
<pre wrap=""><!---->
_______________________________________________
--Bandwidth and Colocation Provided by <a class="moz-txt-link-freetext" href="http://www.api-digital.com">http://www.api-digital.com</a>--
asterisk-dev mailing list
To UNSUBSCRIBE or update options visit:
<a class="moz-txt-link-freetext" href="http://lists.digium.com/mailman/listinfo/asterisk-dev">http://lists.digium.com/mailman/listinfo/asterisk-dev</a>
</pre>
</blockquote>
<br>
</body>
</html>