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<DIV><FONT face=Arial></FONT> </DIV>
<DIV><FONT face=Arial size=2>I don't believe that this issue is with Asterisk.
The issue is that the phone does not set up the RTP stream until it receives the
"200 OK". Asterisk sets up the RTP stream when it receives, or sends, the
message with SDP (either INVITE message, 180 response or 183 response), as per
the RFC.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>The firmware for the 91XXi phones were branched
from the 480i version back in late 2004. Since this branching, the 480i firmware
has been fixed, and the phone now sets up the "early media".</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>The following is from the "KNOWN ISSUES" section of
the 480i release notes from November 2004, v1.0.0.50.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial><FONT size=2>> 2.6 FAILURE TO PROCESS
"REMOTE RINGBACK"<BR>> A device may send
an invitation that includes an SDP message. When the terminator responds with an
alert message, it many also contain SDP message. This is known as "early media"
or "remote ring-back" and indicates that >
the terminator will provide ring-back tone. The
originating device should provide this tone to the originating user.
Currently, the firmware does not process this remote
ringback.<BR>> Currently under
investigation.<BR></FONT></DIV></FONT>
<DIV><FONT face=Arial size=2>I would suspect that the 91XXi firmware was never
updated to correct this problem.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Also, The Asterisk voicemail application will
answer the channel if it hasn't already been answered, before playing any
prompts. Answering the channel send a "200 OK" message to the phone. The phone
will then set up the RTP voice stream for the call. So, if the phone is
"clipping" the begining of the voice prompts, it is an indication that the phone
is taking a long time to set up the RTP after receiving the "200 OK"
message.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>This behaviour is not seen on the other call
servers that Aastra test the phone against, due to the fact that this issue was
known a long time ago, and these call servers have incorporated a workaround to
deal with it.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Fortunately, there is a workaround for Asterisk as
well. In your dialplan, issue an "Answer" and "wait(1)" berfore sending the call
to voicemail.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Regards,</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Derek</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<BLOCKQUOTE
style="PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #000000 2px solid; MARGIN-RIGHT: 0px">
<DIV style="FONT: 10pt arial">----- Original Message ----- </DIV>
<DIV
style="BACKGROUND: #e4e4e4; FONT: 10pt arial; font-color: black"><B>From:</B>
<A title=fwebb@imminc.com href="mailto:fwebb@imminc.com">Franklin Webb</A>
</DIV>
<DIV style="FONT: 10pt arial"><B>To:</B> <A
title=Asterisk-Users@lists.digium.com
href="mailto:Asterisk-Users@lists.digium.com">Asterisk-Users@lists.digium.com</A>
</DIV>
<DIV style="FONT: 10pt arial"><B>Sent:</B> Friday, August 19, 2005 12:49
PM</DIV>
<DIV style="FONT: 10pt arial"><B>Subject:</B> [Asterisk-Users] Asterisk not
conforming to the RFC?/Aastra phonedelay issue</DIV>
<DIV><BR></DIV>
<DIV><FONT face=Arial size=2>Fellow list members,</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>I have run into an issue where I encounter a
delay at the beginning of a phone conversation when I make outgoing calls
through Asterisk with an Aastra 9133i hardphone. This is most noticable
when I call a voicemail system with the Aasta and then with a land line or
other VoIP phone. The first word or two of the voicemail message is
generally cut off.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>According to Aastra's engineering this is because
Asterisk does not confromt o the RFC, setting FTP voice stream before getting
the ACK. They have not seen this with other call servers besides
Asterisk.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Has anyone else seen this sort of behaviour or is
aware of this?</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Right now we are in the process of switching over
the business edition, and we are wondering if we will see a difference in this
problem.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Thanks,</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Franklin Webb</FONT></DIV>
<P>
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