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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'>I am noticing that calls that come from our
IAX pstn gateway provider and terminate to our Asterisk IVR do not receive
ringing when an extension is dialed. For example: <o:p></o:p></span></font></p>

<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<ol style='margin-top:0in' start=1 type=1>
 <li class=MsoNormal style='color:navy;mso-list:l0 level1 lfo1'><font size=2
     color=navy face=Arial><span style='font-size:10.0pt;font-family:Arial'>An
     inbound PSTN caller calls our number <o:p></o:p></span></font></li>
 <li class=MsoNormal style='color:navy;mso-list:l0 level1 lfo1'><font size=2
     color=navy face=Arial><span style='font-size:10.0pt;font-family:Arial'>Asterisk
     answers and provides greeting</span></font> <font size=2 face=Arial><span
     style='font-size:10.0pt;font-family:Arial'><o:p></o:p></span></font></li>
 <li class=MsoNormal style='color:navy;mso-list:l0 level1 lfo1'><font size=2
     color=navy face=Arial><span style='font-size:10.0pt;font-family:Arial'>PSTN
     user dials extension of internal SIP phone</span></font> <font size=2
     face=Arial><span style='font-size:10.0pt;font-family:Arial'><o:p></o:p></span></font></li>
 <li class=MsoNormal style='color:navy;mso-list:l0 level1 lfo1'><font size=2
     color=navy face=Arial><span style='font-size:10.0pt;font-family:Arial'>No ringback
     is heard from PSTN callers perspective</span></font> <font size=2
     face=Arial><span style='font-size:10.0pt;font-family:Arial'><o:p></o:p></span></font></li>
 <li class=MsoNormal style='color:navy;mso-list:l0 level1 lfo1'><font size=2
     color=navy face=Arial><span style='font-size:10.0pt;font-family:Arial'>SIP
     user picks up or the voicemail picks up in the PSTN caller didn&#8217;t
     get frustrated and hang up due to silence</span></font> <font size=2
     face=Arial><span style='font-size:10.0pt;font-family:Arial'><o:p></o:p></span></font></li>
</ol>

<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>On the other hand&#8230;If I call <a
href="mailto:info@ourcompany.com">info@ourcompany.com</a> from an external sip
phone all is well as far as ringing goes.<o:p></o:p></span></font></p>

<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'><o:p>&nbsp;</o:p></span></font></p>

<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>I added an &#8220;r&#8221; to the dial command to try to
force the issue with no success.<o:p></o:p></span></font></p>

<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'><o:p>&nbsp;</o:p></span></font></p>

<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>This is what asterisk says:<o:p></o:p></span></font></p>

<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>== CDR updated on IAX2/sixTel@sixTel/2<o:p></o:p></span></font></p>

<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>&nbsp;&nbsp;&nbsp; -- Executing
Dial(&quot;IAX2/sixTel@sixTel/2&quot;, &quot;SIP/101|20|r&quot;) in new stack<o:p></o:p></span></font></p>

<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>&nbsp;&nbsp;&nbsp; -- Called 101<o:p></o:p></span></font></p>

<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>&nbsp;&nbsp;&nbsp; -- SIP/101-178a is
ringing<o:p></o:p></span></font></p>

<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>However, I&#8217;m still not getting any
tone. If you have any other ideas let me know. (I believe the &#8220;r&#8221;
suggests that it should be sending back tone but my caller is not getting any)
BTW I&#8217;m using the latest stable build 1.0.2 of Asterisk.<o:p></o:p></span></font></p>

<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>Thanks, Chad<o:p></o:p></span></font></p>

<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'><o:p>&nbsp;</o:p></span></font></p>

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