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<p class="MsoNormal"><span style="font-size:9.0pt;font-family:"Verdana","sans-serif"">Let’s see if I can do this, so something like this?<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:9.0pt;font-family:"Verdana","sans-serif"">Using the PSTN Source IP to decide on the asterisk peer?<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:9.0pt;font-family:"Verdana","sans-serif"">Not really sure on what variable dispatch will return as it’s chosen ip?<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:9.0pt;font-family:"Verdana","sans-serif""><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="font-size:9.0pt;font-family:"Verdana","sans-serif"">IP_auth ->Dispatch->Force Socket->Relay<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:9.0pt;font-family:"Verdana","sans-serif""><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="font-size:9.0pt;font-family:"Verdana","sans-serif"">route{<o:p></o:p></span></p>
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<o:p></o:p></span></p>
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<p class="MsoNormal"><span style="font-size:9.0pt;font-family:"Verdana","sans-serif""> # load-balance dispatching on gateways group '1'<o:p></o:p></span></p>
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<p class="MsoNormal"><span style="font-size:9.0pt;font-family:"Verdana","sans-serif""> xlog("L_DBG", "--- SCRIPT: going to <$ru<span style="color:red">> via <$du>\n");</span><o:p></o:p></span></p>
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<p class="MsoNormal"><span style="font-size:9.0pt;font-family:"Verdana","sans-serif""> force_send_socket($sndto:5070);<o:p></o:p></span></p>
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<p class="MsoNormal"><span style="font-size:9.0pt;font-family:"Verdana","sans-serif""> return;<o:p></o:p></span></p>
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<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:9.0pt;font-family:"Verdana","sans-serif""><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="font-size:9.0pt;font-family:"Verdana","sans-serif"">Most definitely, thank you for your time.<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:9.0pt;font-family:"Verdana","sans-serif""><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="font-size:9.0pt;font-family:"Verdana","sans-serif"">Matt Scott<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:9.0pt;font-family:"Verdana","sans-serif""><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""> sr-users-bounces@lists.sip-router.org [mailto:sr-users-bounces@lists.sip-router.org]
<b>On Behalf Of </b>Olle E. Johansson<br>
<b>Sent:</b> Friday, January 18, 2013 1:09 AM<br>
<b>To:</b> SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List<br>
<b>Subject:</b> Re: [SR-Users] Invite - negotiate to asterisk as peer not kamailio<o:p></o:p></span></p>
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<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal">17 jan 2013 kl. 18:07 skrev "Scott, Matt" <<a href="mailto:mscott@homeadvisor.com">mscott@homeadvisor.com</a>>:<o:p></o:p></p>
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<p class="MsoNormal"><span style="font-size:9.0pt;font-family:"Verdana","sans-serif"">Pstn->Kamailio->dispatcher->asterisk</span><o:p></o:p></p>
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<p class="MsoNormal"><span style="font-size:9.0pt;font-family:"Verdana","sans-serif""> </span><o:p></o:p></p>
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<p class="MsoNormal"><span style="font-size:9.0pt;font-family:"Verdana","sans-serif""> </span><o:p></o:p></p>
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<p class="MsoNormal"><span style="font-size:9.0pt;font-family:"Verdana","sans-serif"">When calls come in, they are sent to asterisk, and the peer is negotiated as Kamailio.</span><o:p></o:p></p>
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<p class="MsoNormal"><span style="font-size:9.0pt;font-family:"Verdana","sans-serif"">How can I have different peer settings, for example dtmf?</span><o:p></o:p></p>
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<p class="MsoNormal">I have an Asterisk branch that separates peers beyond the proxy (pinetree-1.4) based on the via received headers.<o:p></o:p></p>
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<p class="MsoNormal">I have it in production for Asterisk 1.4 in multiple places, but don't know the current state in relationship to<o:p></o:p></p>
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<p class="MsoNormal">latest source code.<o:p></o:p></p>
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<o:p></o:p></p>
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<p class="MsoNormal"><span style="font-size:9.0pt;font-family:"Verdana","sans-serif""> </span><o:p></o:p></p>
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<p class="MsoNormal"><span style="font-size:9.0pt;font-family:"Verdana","sans-serif"">I’m thinking I want the call to come in, then when the invite is sent to asterisk, the invite should be from the pstn-peer and not Kamailio, make sense?</span><o:p></o:p></p>
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<p class="MsoNormal"><span style="font-size:9.0pt;font-family:"Verdana","sans-serif"">Then I can have separate peers, with their own dtmf/audio settings, but I see the invite as:</span><o:p></o:p></p>
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<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal">Remember that Asterisk first match incoming calls on users with the FROM header. You can separate based on the From header. But for PSTN, that doesn't really help...<o:p></o:p></p>
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<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal">If you have a limited set of profiles you can add port numbers to Asterisk peers, like have a peer on port 5060 and another on the kamailio IP on 5065. Have kamailio open both sockets and use force_socket to send from the proper one.<o:p></o:p></p>
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<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal">/O<o:p></o:p></p>
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<o:p></o:p></p>
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<p class="MsoNormal"><span style="font-size:9.0pt;font-family:"Verdana","sans-serif""> </span><o:p></o:p></p>
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<p class="MsoNormal"><span style="font-size:9.0pt;font-family:"Verdana","sans-serif"">INVITE<span class="apple-converted-space"> </span><a href="sip:+1866NXXNXXX@%3ckamailio"><span style="color:purple">sip:+1866NXXNXXX@<kamailio</span></a><span class="apple-converted-space"> </span>ip>:5060
SIP/2.0</span><o:p></o:p></p>
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<p class="MsoNormal"><span style="font-size:9.0pt;font-family:"Verdana","sans-serif"">Via: SIP/2.0/UDP <Kamailio ip>;branch=z9hG4bKc087.12db3c91.0</span><o:p></o:p></p>
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<p class="MsoNormal"><span style="font-size:9.0pt;font-family:"Verdana","sans-serif"">Via: SIP/2.0/UDP <pstn-peer ip>:5060;branch=z9hG4bK0cB824c55100a4f2b8b</span><o:p></o:p></p>
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<p class="MsoNormal"><span style="font-size:9.0pt;font-family:"Verdana","sans-serif"">From: <sip:+1NXXNXXXXXXXX@<pstn-peer ip>:5060;isup-oli=0>;tag=gK0c731df8</span><o:p></o:p></p>
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<p class="MsoNormal"><span style="font-size:9.0pt;font-family:"Verdana","sans-serif"">To: <sip:+1NXXNXXNXXX@<kamailio ip>:5060></span><o:p></o:p></p>
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<p class="MsoNormal"><span style="font-size:9.0pt;font-family:"Verdana","sans-serif"">Call-ID: 1024240378_131584110@<pstn-peer ip></span><o:p></o:p></p>
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<p class="MsoNormal"><span style="font-size:9.0pt;font-family:"Verdana","sans-serif""> </span><o:p></o:p></p>
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<p class="MsoNormal"><span style="font-size:9.0pt;font-family:"Verdana","sans-serif"">Any help or suggestions are very welcome..</span><o:p></o:p></p>
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<p class="MsoNormal"><span style="font-size:9.0pt;font-family:"Verdana","sans-serif""> </span><o:p></o:p></p>
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<p class="MsoNormal"><b><span style="font-size:8.5pt;font-family:"Helvetica","sans-serif";
color:#333333">Matt Scott</span></b><o:p></o:p></p>
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<p class="MsoNormal"> <o:p></o:p></p>
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<p class="MsoNormal"><span style="font-size:13.5pt;font-family:"Helvetica","sans-serif"">_______________________________________________<br>
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list<br>
<a href="mailto:sr-users@lists.sip-router.org"><span style="color:purple">sr-users@lists.sip-router.org</span></a><br>
<a href="http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users"><span style="color:purple">http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users</span></a><o:p></o:p></span></p>
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<p class="MsoNormal"><o:p> </o:p></p>
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