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<TITLE>Re: [SR-Users] Hi Daniel.</TITLE>
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<FONT FACE="Calibri, Verdana, Helvetica, Arial"><SPAN STYLE='font-size:11pt'>I am no Daniel, but here is a stab at it ...<BR>
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The pseudo variable “oP” maybe something you can do a quick if/else on?<BR>
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Taken from <a href="http://www.kamailio.org/dokuwiki/doku.php/pseudovariables:1.5.x">http://www.kamailio.org/dokuwiki/doku.php/pseudovariables:1.5.x</a> :<BR>
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</SPAN></FONT><FONT SIZE="4"><FONT FACE="Helvetica, Verdana, Arial"><SPAN STYLE='font-size:13pt'><B>Transport protocol of SIP request original URI<BR>
</B></SPAN></FONT></FONT><FONT FACE="Helvetica, Verdana, Arial"><B><FONT SIZE="2"><SPAN STYLE='font-size:10pt'>$oP</SPAN></FONT></B><FONT SIZE="2"><SPAN STYLE='font-size:10pt'> - reference to transport protocol of original R-URI<BR>
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There is also dP and rP as well.<BR>
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Hope this helps,<BR>
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-graham<BR>
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On 3/16/12 12:11 PM, "Andres Collazos" <<a href="anfecora@gmail.com">anfecora@gmail.com</a>> wrote:<BR>
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</SPAN></FONT><BLOCKQUOTE><FONT FACE="Calibri, Verdana, Helvetica, Arial"><SPAN STYLE='font-size:11pt'>thanks for all the support for all this years.<BR>
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Can you please help me to know if there is any way to route sip calls based on transport protocol, for example a call incoming on tcp i will assign a route and if a call comes in udp i will assign a different route.<BR>
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my scenario is calls coming from different devices registering to kamailio and then kamailio send those calls to asterisk.<BR>
unfortunately i have to create a different peer set for each device. for this scenario i have two types of UAs and they need completely different configuration on the asterisk switch.<BR>
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i am planing on segregate the traffic and build a media server for each type of device means 2 asterisk, teh only way that i can identify those incoming registrations is by the use of the port one client connects trough udp the other trough tcp.<BR>
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I appreciated any input in this matter.<BR>
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and again thank a lot to all for the great support.<BR>
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Andres Collazos.<BR>
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