<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
<html>
<head>
<meta http-equiv="Content-Type" content="text/html;charset=ISO-8859-1">
<title></title>
</head>
<body text="#000000" bgcolor="#ffffff">
<pre wrap="">Thanks for the answer. Doesn't solve my problem, but that's only because I didn't state my goal. You have corrected a misconseption on my part, which ought to get me closer. I'll explain...
Indeed, I do have the "tT" options in the dial command. This is because I thought this would enable the use of the '#' for transfers, and it works satisfactorily. I also have various '*N' definitions in features.conf, but these don't work. I suppose I do have to rethink my strategy as you've suggested, but I don't know how to have my cake and eat it.. (?)
By the way, I am using various SIP phones, with various DTMF detection techniques (e.g. ZyXEL wifi:inband, Grandstream BT101 and ATA-488:INFO) with apparent success because many features do work (such as transfer with #).
Message: 22
Date: Sun, 30 Oct 2005 10:57:57 -0400
From: Andrew Kohlsmith <a class="moz-txt-link-rfc2396E"
href="mailto:akohlsmith-asterisk@benshaw.com"><akohlsmith-asterisk@benshaw.com></a>
Subject: Re: [Asterisk-Users] gotta be a dumb question...
To: <a class="moz-txt-link-abbreviated"
href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a>
Message-ID: <a class="moz-txt-link-rfc2396E"
href="mailto:200510300957.57769.akohlsmith-asterisk@benshaw.com"><200510300957.57769.akohlsmith-asterisk@benshaw.com></a>
Content-Type: text/plain; charset="iso-8859-1"
On Sunday 30 October 2005 09:44, Bill Michaelson wrote:
</pre>
<blockquote type="cite">
<pre wrap=""><span class="moz-txt-citetags">> </span>-- Attempting native bridge of SIP/215-b09e and SIP/259412-5967
<span class="moz-txt-citetags">></span>
<span class="moz-txt-citetags">> </span>Now, I've got canreinvite=no in every sip definition, but it happens
<span class="moz-txt-citetags">> </span>anyway.
</pre>
</blockquote>
<pre wrap=""><!---->
That has nothing to do with reinvites.
In Asterisk terms, a native bridge between two channels is the lowest-latency
connection between those channels without dropping out of the loop entirely.
Essentially a native bridge just reads voice frames from one and transmits
them to the other. There is no codec translation or any other goodness going
on.
When you hit a DTMF digit (you must be using inband DTMF here I think), the
native bridge must be dropped because Asterisk needs to prepare to do
something with the DTMF (transfer, etc.) -- when Asterisk has determined that
it doesn't need to do anything special, it sets up the native bridge again to
minimize the latency once again.
The fact that your * is getting "swallowed" tells me that you are using * in
features.conf to denote special keypresses to Asterisk. In Dial() you likely
have the 't' or 'T' flags set, which causes Asterisk to "think" that those
DTMF digits are for it, not for the other side. Either edit features.conf,
remove the 't' or 'T' flags from the Dial() command or rethink your strategy.
I hope this is an acceptable answer, and I certainly hope it's accurate. It's
my understanding of the system anyway. <img
src="cid:part1.02080606.03000501@cosi.com" alt=":-)"
class="moz-txt-smily" height="19" width="19" align="middle"> If you prefer not to have these
types of messages, you need to turn DOWN the verbosity level.
-A.
</pre>
</body>
</html>