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<font size="-1"><font face="Verdana">You may need to add "r" as
option perameter to dial command.</font></font><br>
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<title>Signatures <a class="moz-txt-link-abbreviated" href="mailto:faisal@vopium.com">faisal@vopium.com</a></title>
<p style="font-family: Verdana;" class="MsoNormal"><small><small><small><small><small><small><small><small><span
style="font-size: 10pt;">Regards,<o:p></o:p></span></small></small></small></small></small></small></small></small></p>
<p style="font-family: Verdana;" class="MsoNormal"><small><small><small><small><small><small><small><small><span
style="font-size: 10pt;"><o:p></o:p>Faisal
Hanif</span></small></small></small></small></small></small></small></small><br>
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On 7/26/2010 9:39 PM, Chris Ramirez wrote:
<blockquote cite="mid:4C4DBA5D.2090501@tele-onecom.com" type="cite">
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The problem we are having with Asterisk is when we initiate a call
via
a Zap line and it goes out on a Sip line. When it goes out via Sip
we
hear no sound until the party we are calling answers the line. If
the
call were to go out Sip-Sip or Zap-Zap it works perfectly fine. It
is
only with the Zap-Sip calls. If anyone knows anything that could
possibly help it would be greatly appreciated. I have checked many
different things already and tried comparing Zap-Zap and Zap-Sip
call
logs. Thanks!<br>
<div class="moz-signature">-- <br>
<font color="gray"><b>Chris Ramirez</b></font>
<br>
<font color="green">TELE-ONE COMMUNICATIONS, INC.</font>
<br>
<a moz-do-not-send="true" class="moz-txt-link-abbreviated"
href="mailto:cramirez@tele-onecom.com">cramirez@tele-onecom.com</a>
<br>
903-531-0777
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