Thanks. Indeed ringing instead of MoH "which was missing files" fixed the issue. Thanks for the quick great tip. Simple things hide from us sometime.<br><br><div class="gmail_quote">On Wed, Feb 16, 2011 at 3:48 PM, Danny Nicholas <span dir="ltr"><<a href="mailto:danny@debsinc.com">danny@debsinc.com</a>></span> wrote:<br>
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<p class="MsoNormal"><b><font size="2" face="Tahoma"><span style="font-size:10.0pt;font-family:Tahoma;font-weight:bold">From:</span></font></b><font size="2" face="Tahoma"><span style="font-size:10.0pt;font-family:Tahoma">
<a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank">asterisk-users-bounces@lists.digium.com</a>
[mailto:<a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank">asterisk-users-bounces@lists.digium.com</a>] <b><span style="font-weight:bold">On Behalf Of </span></b>Bruce B<br>
<b><span style="font-weight:bold">Sent:</span></b> Wednesday, February 16, 2011
2:33 PM<br>
<b><span style="font-weight:bold">To:</span></b> Asterisk
Users Mailing List - Non-Commercial Discussion<br>
<b><span style="font-weight:bold">Subject:</span></b> [asterisk-users] No ring
tone on inbound call - but channelconnects fine</span></font></p>
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<p class="MsoNormal"><font size="3" face="Times New Roman"><span style="font-size:12.0pt">Hi Everyone,</span></font></p>
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<p class="MsoNormal"><font size="3" face="Times New Roman"><span style="font-size:12.0pt">I have a SIP turnk which works fine with both inbound and outbound
calling. However, the only issue is that there is no Ring Tone if someone calls
us. The phones used are Aastra and Polycom connected to the PBX via VPN (SIP).</span></font></p>
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<p class="MsoNormal"><font size="3" face="Times New Roman"><span style="font-size:12.0pt">I do get an outbound ring tone, so it's not that there is any media
loss between the phones and the PBX. But when the DID is called there is dead
silence until the call is picked up.</span></font></p>
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<p class="MsoNormal"><font size="3" face="Times New Roman"><span style="font-size:12.0pt">What is generally causing something like this? and where should I start
looking? Much appreciate your experienced tips.</span></font></p>
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<p class="MsoNormal"><font size="3" face="Times New Roman"><span style="font-size:12.0pt">Thanks</span></font></p>
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<p class="MsoNormal"><font size="2" color="navy" face="Arial"><span style="font-size:10.0pt;font-family:Arial;color:navy">This sounds like a dialplan problem. My
thought is that your SIP trunk should go to an “incoming” context
that does something like this:</span></font></p>
<p class="MsoNormal"><font size="2" color="navy" face="Arial"><span style="font-size:10.0pt;font-family:Arial;color:navy">In-house phones are 1000 and 1001</span></font></p>
<p class="MsoNormal"><font size="2" color="navy" face="Arial"><span style="font-size:10.0pt;font-family:Arial;color:navy">[incoming]</span></font></p>
<p class="MsoNormal"><font size="2" color="navy" face="Arial"><span style="font-size:10.0pt;font-family:Arial;color:navy">Exten => s,1,answer</span></font></p>
<p class="MsoNormal"><font size="2" color="navy" face="Arial"><span style="font-size:10.0pt;font-family:Arial;color:navy">Exten =>
s,n,Dial(SIP/1000&SIP/1001,30,mKkTt)</span></font></p>
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