<div>Hi Shaun and Robb,</div>
<div> </div>
<div>I tried the Avaya IP small Office with Lucent Analog phones. it worked fine on hang ups - i think it is my old analog phone is the root cause.</div>
<div> </div>
<div>I have only one major issue now.</div>
<div> </div>
<div>I am not getting the Caller ID Name and Caller ID number from avaya to asterisk. </div>
<div> </div>
<div>Can you provide me your valuable input.</div>
<div> </div>
<div>This is what i have.</div>
<div>When i configured the SIP trunk, </div>
<div> </div>
<div>Under SIP line - i had </div>
<div>Primary authentication name = avayanew</div>
<div>
<div>Primary authentication Password = avayanew</div>
<div> </div>
<div>and also under the SIP URI:</div>
<div>Local URI, Contact and Display name - i had selected "use authentication name" for successful calls, but as said caller ID is not passed through asterisk.</div>
<div> </div>
<div>when i try the "use user data" in there, i get "TTel:" the problem i had before and cannot make/receive calls.</div></div>
<div> </div>
<div>Please advise</div>
<div> </div>
<div>Thanks as always</div>
<div> </div>
<div>Regards</div>
<div>Krishna<br></div>
<div class="gmail_quote">On Mon, Nov 10, 2008 at 6:57 PM, 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 Tue, Nov 11, 2008 at 4:56 AM, Krishna Sumanth Chava<br><<a href="mailto:kschava@gmail.com">kschava@gmail.com</a>> wrote:<br>> HI Shaun and Robb,<br>><br>> Thanks for the assistance.<br>
><br>> I was able to force the codecs and had avaya talk in the right way. Also<br>> addressed the DTMF issues.<br><br></div>Glad to hear it.<br>
<div class="Ih2E3d"><br>> I seem to be having issues with asterisk and avaya not detecting Hang ups.<br>> i am using the Analog phones connected to the POTS ports on the IP Office. I<br>> will try connecting the avaya Analog and Avaya IP Phone to IP Office and see<br>
> if that makes any difference.<br><br></div>What does SSA show when one end has hung up? If it still shows the<br>call as active, then a disconnect signal has gone missing.<br><br>I've never experienced this problem, but then again the only thing we<br>
use the POTS ports for is faxing and this is forced to use our PRI<br>circuits. All of our handsets including conference room phones are IP.<br>
<div>
<div></div>
<div class="Wj3C7c"><br>-Shaun<br><br>_______________________________________________<br>-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com/" target="_blank">http://www.api-digital.com</a> --<br>
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