<div dir="ltr">Hi Bogdan, Leon <br><div><br></div><div>Thank you very much for your information<br></div><div><br></div><div>I have some questions about failover configuration.<br><br></div><div>I have two IP addresses of the Cisco redundancy SIP Trunking servers. The OpenSIPS can ping and detect which server is available before perform the routing and making call to do the fail-over situation, can you provide me some script examples for this situation. Thanks<br><br><br></div><div>My Sip Trunk configuration is :<br></div><div><br>The sip trunk configuration would be like below.<br> # all numbers starting with 55 are to be sent to CUCM<br> if ($rU =~ "^55[0-9]+$") {<br> # replace the domain part of RURI to point to CUCM<br> rewritehostport("CUCM_IP:CUCM_PORT");<br> # route the call out based on RURI<br> route(1);<br> }<br><br><br></div><div>Thanks <br></div><div>Kind Regards, <br><br></div><div>Steven <br></div><div><br></div><div class="gmail_extra"><br><div class="gmail_quote">On 12 January 2011 at 13:18, Leon Li <span dir="ltr"><<a href="mailto:Leon.Li@aarnet.edu.au" target="_blank">Leon.Li@aarnet.edu.au</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div link="blue" vlink="purple" lang="EN-AU"><div><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">Hi Steven,<u></u><u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"><u></u> <u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">To configure the trunk in CUCM, go to Device > Trunk, add a new “SIP trunk”.<u></u><u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"><u></u> <u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">The configuration fields are pretty straight forward. Important ones are<u></u><u></u></span></p><p><u></u><span style="font-size:11.0pt;font-family:Symbol;color:#1f497d"><span>·<span style="font:7.0pt "Times New Roman""> </span></span></span><u></u><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">Destination Address, i.e. opensips IP<u></u><u></u></span></p><p><u></u><span style="font-size:11.0pt;font-family:Symbol;color:#1f497d"><span>·<span style="font:7.0pt "Times New Roman""> </span></span></span><u></u><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">Port, if not 5060<u></u><u></u></span></p><p><u></u><span style="font-size:11.0pt;font-family:Symbol;color:#1f497d"><span>·<span style="font:7.0pt "Times New Roman""> </span></span></span><u></u><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">CSS for inbound and outbound calls. (this decide what number you can send calls to and receive calls from opensips)<u></u><u></u></span></p><p><u></u><span style="font-size:11.0pt;font-family:Symbol;color:#1f497d"><span>·<span style="font:7.0pt "Times New Roman""> </span></span></span><u></u><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">Any number transformation if you have<u></u><u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"><u></u> <u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">This is the basic. If you have questions about particular fields, please mail in details.<u></u><u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"><u></u> <u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">Regards,<u></u><u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">Leon<u></u><u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"><u></u> <u></u></span></p><div style="border:none;border-top:solid #b5c4df 1.0pt;padding:3.0pt 0cm 0cm 0cm"><p class="MsoNormal"><b><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif"" lang="EN-US">From:</span></b><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif"" lang="EN-US"> <a href="mailto:users-bounces@lists.opensips.org" target="_blank">users-bounces@lists.opensips.org</a> [mailto:<a href="mailto:users-bounces@lists.opensips.org" target="_blank">users-bounces@lists.opensips.org</a>] <b>On Behalf Of </b>steven chew<br><b>Sent:</b> Tuesday, 11 January 2011 11:50 AM<br><b>To:</b> OpenSIPS users mailling list<br><b>Subject:</b> Re: [OpenSIPS-Users] How to implement a SIP Trunk in between twoSIP servers.<u></u><u></u></span></p></div><div><div class="h5"><p class="MsoNormal"><u></u> <u></u></p><p class="MsoNormal">Hi Bogdan,<u></u><u></u></p><div><p class="MsoNormal"><u></u> <u></u></p></div><div><p class="MsoNormal">Thanks for your reply.<u></u><u></u></p></div><div><p class="MsoNormal"><u></u> <u></u></p></div><div><p class="MsoNormal"><u></u> <u></u></p></div><div><p class="MsoNormal">Your script is very useful for calling between two opensips servers which I have tested.<br><br>However, I don't know how to configure on CUCM 7.0 which I am using.<br><br>At the moment, CUCM 7.0 is using Web Config via the Web Browser. <br><br>Can you let me know how to configure on CUCM 7.0?<br><br>I will appreciate very much if you give some instructions for configuring SIP Trunk on CUCM7.0<u></u><u></u></p></div><div><p class="MsoNormal"><u></u> <u></u></p></div><div><p class="MsoNormal"><u></u> <u></u></p></div><div><p class="MsoNormal">Thanks<br>Kind regards,<u></u><u></u></p></div><div><p class="MsoNormal" style="margin-bottom:12.0pt">Steven,<u></u><u></u></p><div><p class="MsoNormal">On 10 January 2011 19:33, Bogdan-Andrei Iancu <<a href="mailto:bogdan@voice-system.ro" target="_blank">bogdan@voice-system.ro</a>> wrote:<u></u><u></u></p><p class="MsoNormal">Hi Steven,<br><br>To do that, you need to add in opensips some routing to 1) recognize the numbers that needs to be sent to CUCM and 2)route that calls to CUCM.<br><br>For script logic it sounds like : if you receive a new call (initial INVITE) for your local domain, check the URI and divert. If you look at the default config file, there is comment "# requests for my domain" -> from that point further you have only initial INVITEs for your local domain, so you can add after:<br><br> # all numbers starting with 55 are to be sent to CUCM<br> if ($rU =~ "^55[0-9]+$") {<br> # replace the domain part of RURI to point to CUCM<br> rewritehostport("CUCM_IP:CUCM_PORT");<br> # route the call out based on RURI<br> route(1);<br> }<br><br><br>For the other way around, you have to put a similar logic in CUCM, like to divert all calls starting with "12" to opensips - and replace the domain on RURI with the IP/domain of opensips.<u></u><u></u></p><div><p class="MsoNormal"><br><br>Regards,<br>Bogdan<br><br>steven chew wrote:<u></u><u></u></p></div><blockquote style="border:none;border-left:solid #cccccc 1.0pt;padding:0cm 0cm 0cm 6.0pt;margin-left:4.8pt;margin-right:0cm"><div><p class="MsoNormal">Hi Bogdan,<br><br>Thank you very much for your reply.<br><br>I have an Opensips Server and a Cisco Unified Communication Manager (CUCM).<br><br>If I want to make calls from Opensips Server to CUCM and CUCM to Opensips Server.<br><br>For example:<br>1) If I dial an extension number "5566" from a SIP Phone "12345" under Opensips Server, it will try to call to a Cisco IP Phone "5566" from CUCM through a SIP Trunk.<br>2) If I dial an extension number "12345" from a Cisco IP Phone "5566" under CUCM, it will try to call to a SIP Phone "12345" under Opensips Server through a SIP Trunk.<br><br>Can you give some instructions how to configure the above scenario for dialing extension numbers?<br><br>Thanks<br>Steven, <u></u><u></u></p></div><div><div><p class="MsoNormal">On 6 January 2011 21:31, Bogdan-Andrei Iancu <<a href="mailto:bogdan@voice-system.ro" target="_blank">bogdan@voice-system.ro</a> <mailto:<a href="mailto:bogdan@voice-system.ro" target="_blank">bogdan@voice-system.ro</a>>> wrote:<br><br> Hi Steven,<br><br> If you use the opensips default script, your opensips will accept<br> calls from any other external SIP entities (call targeting a local<br> opensips subscriber).<br><br> If you want to configure your opensips to accept foreign calls<br> only form a specific IP address, you can use the permission<br> module, with address table to implement IP-based authentication.<br><br> Best regards,<br> Bogdan<br><br> steven chew wrote:<br><br> Hi everyone,<br><br> I am a newbie with SIP-Trunk in OpenSips.<br> I have a Cisco Communication Unified Manager and a OpenSips<br> Server running in two different Virtual Machines.<br><br> I would like to have a SIP trunk in between them "Cisco<br> Communication Unified Manager and OpenSips Server".<br> Therefore, I can make a call from OpenSips Server's SIP<br> Clients to Cisco IP Phone.<br> What should I need to add into opensips.cfg configuration file?<br><br> I hope you can give some simple examples how to do it.<br> I look forward to hearing from your advise asap.<br><br> Thanks<br> Regards,<br> -Steven.<br><br> ------------------------------------------------------------------------<br><br> _______________________________________________<br> Users mailing list<u></u><u></u></p></div></div><p class="MsoNormal"> <a href="mailto:Users@lists.opensips.org" target="_blank">Users@lists.opensips.org</a> <mailto:<a href="mailto:Users@lists.opensips.org" target="_blank">Users@lists.opensips.org</a>><u></u><u></u></p><div><p class="MsoNormal"><br> <a href="http://lists.opensips.org/cgi-bin/mailman/listinfo/users" target="_blank">http://lists.opensips.org/cgi-bin/mailman/listinfo/users</a><br> <br><br><br> -- Bogdan-Andrei Iancu<br> OpenSIPS Event - expo, conf, social, bootcamp<br> 2 - 4 February 2011, ITExpo, Miami, USA<u></u><u></u></p></div><p class="MsoNormal"> <a href="http://www.voice-system.ro" target="_blank">www.voice-system.ro</a> <<a href="http://www.voice-system.ro" target="_blank">http://www.voice-system.ro</a>><br><br><br> _______________________________________________<br> Users mailing list<br> <a href="mailto:Users@lists.opensips.org" target="_blank">Users@lists.opensips.org</a> <mailto:<a href="mailto:Users@lists.opensips.org" target="_blank">Users@lists.opensips.org</a>><u></u><u></u></p><div><p class="MsoNormal"><br> <a href="http://lists.opensips.org/cgi-bin/mailman/listinfo/users" target="_blank">http://lists.opensips.org/cgi-bin/mailman/listinfo/users</a><br><br><br>------------------------------------------------------------------------<br><br>_______________________________________________<br>Users mailing list<br><a href="mailto:Users@lists.opensips.org" target="_blank">Users@lists.opensips.org</a><br><a href="http://lists.opensips.org/cgi-bin/mailman/listinfo/users" target="_blank">http://lists.opensips.org/cgi-bin/mailman/listinfo/users</a><br> <u></u><u></u></p></div></blockquote><div><div><p class="MsoNormal"><br><br>-- <br>Bogdan-Andrei Iancu<br>OpenSIPS Event - 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