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<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>I am having a similar issue. Asterisk does not show ringback
tone and I investigated this due to it not reading sip invite 180. (or
supposedly not receiving it).. My solution is that now I am using h323<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>(ver 1.4.19)<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>CS<o:p></o:p></span></p>
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color:#1F497D'><o:p> </o:p></span></p>
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<p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span></b><span
style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>
asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] <b>On Behalf Of </b>Yehavi
Bourvine<br>
<b>Sent:</b> Sunday, April 19, 2009 4:22 AM<br>
<b>To:</b> Asterisk Users Mailing List - Non-Commercial Discussion<br>
<b>Subject:</b> Re: [asterisk-users] issue with sip 180 responses<o:p></o:p></span></p>
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<p class=MsoNormal>Maybe it is buffering issues with the kernel? Does it happen
only when there is a peak in the new calls rate? Do all 180 messages get
dropped?<o:p></o:p></p>
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<p class=MsoNormal> <o:p></o:p></p>
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<p class=MsoNormal>
__Yehavi:<o:p></o:p></p>
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<p class=MsoNormal> <o:p></o:p></p>
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<p class=MsoNormal>2009/4/19 Nir Levi <a href="mailto:NirL@bezeqint.co.il">NirL@bezeqint.co.il</a><o:p></o:p></p>
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<p class=MsoNormal><o:p> </o:p></p>
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<p>Hello,<o:p></o:p></p>
<p>SIP invites are accepted from imitator , but 'SIP 180' is not
responded back to imitator. <o:p></o:p></p>
<p> <o:p></o:p></p>
<p>By inspecting the issue , we can *<b>see</b>* the response is generated
and sent from asterisk (via asterisk logger ("sip debug" )) , but
while sniffing the interface with tcpdump, we can't see 180 response on the
interface.<o:p></o:p></p>
<p> <o:p></o:p></p>
<p>We don’t have errors on the interface, firewall is disabled , seems there's
no packet lost (checked with ping with low interval ) , and routes are
ok … <o:p></o:p></p>
<p> <o:p></o:p></p>
<p>By our tests we can see there is a direct connection between the mass of
the calls, and between the lost of sip 180 responses.<o:p></o:p></p>
<p> <o:p></o:p></p>
<p>We're using Asterisk 1.4.4, with real-time configuration, also we made few
*<b>modifications</b>* in asterisk source code (changed <span
style='color:blue'>app_dial.c, /app_macro.c, /func_cdr.c</span>)…<o:p></o:p></p>
<p> <o:p></o:p></p>
<p>We're not sure if the problem on the OS Level (Centos 5.2) or in the
asterisk application.<o:p></o:p></p>
<p>Please assist…<o:p></o:p></p>
<p>~Nir<o:p></o:p></p>
<p> <o:p></o:p></p>
<p> <o:p></o:p></p>
<p> <o:p></o:p></p>
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<p class=MsoNormal><span style='font-size:9.0pt;font-family:"Arial","sans-serif"'>This
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