<div>HI Shaun and Robb,</div>
<div> </div>
<div>Thanks for the assistance. </div>
<div> </div>
<div>I was able to force the codecs and had avaya talk in the right way. Also addressed the DTMF issues.</div>
<div> </div>
<div>I seem to be having issues with asterisk and avaya not detecting Hang ups. i am using the Analog phones connected to the POTS ports on the IP Office. I will try connecting the avaya Analog and Avaya IP Phone to IP Office and see if that makes any difference.</div>
<div> </div>
<div>Thanks again.</div>
<div> </div>
<div>Regards</div>
<div>Krishna<br><br></div>
<div class="gmail_quote">On Mon, Nov 10, 2008 at 12:04 AM, Shaun Ewing <span dir="ltr"><<a href="mailto:sewing@gmail.com">sewing@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid">
<div class="Ih2E3d">On Mon, Nov 10, 2008 at 2:28 PM, Krishna Sumanth Chava<br><<a href="mailto:kschava@gmail.com">kschava@gmail.com</a>> wrote:<br>> Hi Guys,<br>><br>> Thanks that did help to resolve my issue. i tried the ."@<a href="http://10.10.8.1/" target="_blank">10.10.8.1</a>" and it<br>
> worked and i had a successful call but i have the following 2 concerns.<br>><br>> 1. We have voice communication from avaya to asterisk now but avaya<br>> is forcing asterisk to use only codec G723. if i disable G723, it says no<br>
> compatible codecs. While the calls from asterisk to avaya are being accepted<br>> as "alaw"<br><br></div>Make sure you have Compression Mode in your SIP line config on the IP<br>Office set to your desired codec. You'll run into this problem if you<br>
have it set to "Automatic Select".<br><br>Make sure you also reduce the number of codecs on the Asterisk side.<br>For example, our sip.conf entry looks like:<br><br>[ipo-cbr2]<br>type=friend<br>username=ipo-cbr2<br>
secret=xxxxxx<br>host=<a href="http://172.31.2.1/" target="_blank">172.31.2.1</a><br>nat=never<br>context=from-ipo-cbr2<br>insecure=port,invite<br>disallow=all<br>allow=ulaw<br>allow=alaw<br>canreinvite=yes<br>qualify=no<br>
dtmfmode=auto<br><br>To reduce VCM usage, also make sure your IP handsets are using the<br>same codec. If they are, you won't use any VCM channels for a call.<br>
<div class="Ih2E3d"><br>> 2. I am having issues with DTMF. DTMF is not being recognized or being sent<br>> from avaya to asterisk.<br><br></div>It should work. Make sure the Asterisk side has dtmfmode=auto like above.<br>
<font color="#888888"><br>-Shaun<br></font>
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