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<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><font
size=2 color=navy face=Arial><span style='font-size:10.0pt;font-family:Arial;
color:navy'>Guys, I posted the following question on the users group with no
response. Does anyone on this list have any ideas?<o:p></o:p></span></font></p>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><font
size=2 color=navy face=Arial><span style='font-size:10.0pt;font-family:Arial;
color:navy'>--------------------------------------------------------------------<o:p></o:p></span></font></p>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><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> </O:P></span></font><o:p></o:p></p>
<ol start=1 type=1>
<li class=MsoNormal style='color:navy;mso-margin-top-alt:auto;mso-margin-bottom-alt:
auto;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><o:p></o:p></li>
<li class=MsoNormal style='color:navy;mso-margin-top-alt:auto;mso-margin-bottom-alt:
auto;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> <o:p></o:p></li>
<O:P></O:P>
<li class=MsoNormal style='color:navy;mso-margin-top-alt:auto;mso-margin-bottom-alt:
auto;mso-list:l0 level1 lfo1'><font size=2 color=navy face=Arial><span
style='font-size:10.0pt;font-family:Arial'>PSTN user routed through our
IAX pstn provider dials extension of an internal SIP phone</span></font> <o:p></o:p></li>
<O:P></O:P>
<li class=MsoNormal style='color:navy;mso-margin-top-alt:auto;mso-margin-bottom-alt:
auto;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> <o:p></o:p></li>
<O:P></O:P>
<li class=MsoNormal style='color:navy;mso-margin-top-alt:auto;mso-margin-bottom-alt:
auto;mso-list:l0 level1 lfo1'><font size=2 color=navy face=Arial><span
style='font-size:10.0pt;font-family:Arial'>PSTN caller gets frustrated and
hang up due to silence</span></font> <O:P></O:P>if SIP user does not pick
up right away<o:p></o:p></li>
</ol>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><font
size=2 color=navy face=Arial><span style='font-size:10.0pt;font-family:Arial;
color:navy'><O:P>On the other hand…If I call <a
href="mailto:%5bEMAIL%20PROTECTED%5d">[EMAIL PROTECTED]</a> from an external
sip phone all is well as far as ringing goes.<O:P></O:P></span></font><o:p></o:p></p>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><font
size=2 face=Arial><span style='font-size:10.0pt;font-family:Arial'><O:P>I added
an “r” to the dial command to try to force the issue with no
success.<O:P></O:P></span></font><o:p></o:p></p>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><font
size=2 color=navy face=Arial><span style='font-size:10.0pt;font-family:Arial;
color:navy'><O:P>This is what asterisk says:<O:P></O:P></span></font><o:p></o:p></p>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><font
size=2 color=navy face=Arial><span style='font-size:10.0pt;font-family:Arial;
color:navy'><O:P>== CDR updated on IAX2/[EMAIL PROTECTED]/2<O:P></O:P></span></font><o:p></o:p></p>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><font
size=2 color=navy face=Arial><span style='font-size:10.0pt;font-family:Arial;
color:navy'> -- Executing Dial("IAX2/[EMAIL
PROTECTED]/2", "SIP/101|20|r") in new stack<O:P></O:P></span></font><o:p></o:p></p>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><font
size=2 color=navy face=Arial><span style='font-size:10.0pt;font-family:Arial;
color:navy'> -- Called 101<O:P></O:P></span></font><o:p></o:p></p>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><font
size=2 color=navy face=Arial><span style='font-size:10.0pt;font-family:Arial;
color:navy'> -- SIP/101-178a is ringing<O:P></O:P></span></font><o:p></o:p></p>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><font
size=2 color=navy face=Arial><span style='font-size:10.0pt;font-family:Arial;
color:navy'><O:P>However, I’m still not getting any tone. If you have any
other ideas let me know. (I believe the “r” suggests that it should
be sending back tone but my caller is not getting any) BTW I’m using the
latest stable build 1.0.2 of Asterisk.<O:P></O:P></span></font><o:p></o:p></p>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><font
size=2 color=navy face=Arial><span style='font-size:10.0pt;font-family:Arial;
color:navy'><O:P>Thanks, <st1:country-region w:st="on"><st1:place w:st="on">Chad<O:P></O:P></st1:place></st1:country-region>
Brown - IdentityMine</span></font><o:p></o:p></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'><o:p> </o:p></span></font></p>
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