<p style="MARGIN: 0in 0in 0pt" class="MsoNormal"><span style="FONT-FAMILY: 'Calibri','sans-serif'; COLOR: #1f497d; FONT-SIZE: 11pt; mso-ascii-theme-font: minor-latin; mso-hansi-theme-font: minor-latin; mso-bidi-font-family: 'Times New Roman'; mso-bidi-theme-font: minor-bidi; mso-themecolor: dark2">Hi </span></p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal"><span style="FONT-FAMILY: 'Calibri','sans-serif'; COLOR: #1f497d; FONT-SIZE: 11pt; mso-ascii-theme-font: minor-latin; mso-hansi-theme-font: minor-latin; mso-bidi-font-family: 'Times New Roman'; mso-bidi-theme-font: minor-bidi; mso-themecolor: dark2"> </span></p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal"><span style="FONT-FAMILY: 'Calibri','sans-serif'; COLOR: #1f497d; FONT-SIZE: 11pt; mso-ascii-theme-font: minor-latin; mso-hansi-theme-font: minor-latin; mso-bidi-font-family: 'Times New Roman'; mso-bidi-theme-font: minor-bidi; mso-themecolor: dark2">Try usecallerid=no usually the asterisk server will wait two rings before answering when usecallerid is set to yes. </span></p>
<p style="MARGIN: 0in 0in 0pt" class="MsoNormal"><span style="FONT-FAMILY: 'Calibri','sans-serif'; COLOR: #1f497d; FONT-SIZE: 11pt; mso-ascii-theme-font: minor-latin; mso-hansi-theme-font: minor-latin; mso-bidi-font-family: 'Times New Roman'; mso-bidi-theme-font: minor-bidi; mso-themecolor: dark2"> </span></p>
<div style="MARGIN: 0in 0in 0pt" class="MsoNormal"><span style="FONT-FAMILY: 'Calibri','sans-serif'; COLOR: #1f497d; FONT-SIZE: 11pt; mso-ascii-theme-font: minor-latin; mso-hansi-theme-font: minor-latin; mso-bidi-font-family: 'Times New Roman'; mso-bidi-theme-font: minor-bidi; mso-themecolor: dark2">Flavio E. Goncalves</span></div>
<div style="MARGIN: 0in 0in 0pt" class="MsoNormal"><span style="FONT-FAMILY: 'Calibri','sans-serif'; COLOR: #1f497d; FONT-SIZE: 11pt; mso-ascii-theme-font: minor-latin; mso-hansi-theme-font: minor-latin; mso-bidi-font-family: 'Times New Roman'; mso-bidi-theme-font: minor-bidi; mso-themecolor: dark2"><a href="http://www.asteriskguide.com/">www.asteriskguide.com</a></span></div>
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<div class="gmail_quote">2010/4/7 Balu Raman <span dir="ltr"><<a href="mailto:braman09@gmail.com">braman09@gmail.com</a>></span><br>
<blockquote style="BORDER-LEFT: #ccc 1px solid; MARGIN: 0px 0px 0px 0.8ex; PADDING-LEFT: 1ex" class="gmail_quote">Hope some can help me.<br>I have a PSTN coming into TDM400 into Asterisk. We also have direct telephones connected to the PSTN bypassing the Asterisk. When a call comes in on the PSTN the direct connected phones ring first and if no one picks up , Asterisk picks and get routed to internal sip phones. I am not able to find what I should tune to make the calls always go through asterisk without the direct telephones ringing. Things used to work right, suddenly, I have this problem after a recent storm.<br>
Thanks,<br>-braman<br><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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