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</o:shapelayout></xml><![endif]--></head><body lang=EN-US link=blue vlink=purple><div class=WordSection1><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Suppose I have to create a separate gateway xml for each account I’ve had with same sip provider. Upon registration, I should see all these accounts register with same Sip Provider as separate gateways in sofia. As I know that each <b>sip</b> profile is a user agent. A user agent can service exactly one IP:Port. In my case, all my accounts/gateways are register through the same IP:Port to the <b>same sip provider</b>, how does FS keep track of what voip call belongs which account? I am a bit confuse about the concept of multiple gateways share the same IP:port. Would someone enlighten me on this topic?<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'>Cheers,<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Louie<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><div><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"'>From:</span></b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'> Robert Hadley [mailto:robert.hadley@teotech.com] <br><b>Sent:</b> Friday, 25 July 2014 2:33 AM<br><b>To:</b> FreeSWITCH Users Help<br><b>Subject:</b> Re: [Freeswitch-users] multiple sip gateway configuration<o:p></o:p></span></p></div></div><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Hi Louie,<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'>One SIP provider profile (ie. port) may have multiple gateways like we have for our Vitelity SIP trunk.<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'><a href="https://wiki.freeswitch.org/wiki/Provider_Configuration:_Vitelity">https://wiki.freeswitch.org/wiki/Provider_Configuration:_Vitelity</a><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'>Regards,<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Robert<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'><o:p> </o:p></span></p><div><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"'>From:</span></b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'> Louie Liu [<a href="mailto:lliu@multinet.net.au">mailto:lliu@multinet.net.au</a>] <br><b>Sent:</b> Thursday, July 24, 2014 6:02 AM<br><b>To:</b> 'FreeSWITCH Users Help'<br><b>Subject:</b> Re: [Freeswitch-users] multiple sip gateway configuration<o:p></o:p></span></p></div></div><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Thanks your prompt response, maybe I haven’t explain my scenario clearly, Iet me explain:<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>My company would like to setup a voip service to be host in the data centre where Sip_Provider_A and B both have a presence, so avoid voip traffic route across the Internet, but each would need to go through different network interface and gateway to reach the sip providers (see diagram below for illustration). <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><pre> PHONE -> FS(192.168.0.1) -> NAT1(Public IP 1.2.3.4) -> Sip_Provider_A<o:p></o:p></pre><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'> I-</span><span style='font-size:10.0pt;font-family:"Courier New"'>-> NAT2 (Public IP 5.6.7.8) -> Sip_Provider_B</span><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'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>To complicate this even more, Sip_Provider_B offers multiple accounts to cover the existing ISDN connections we’ve got. To me it makes sense to setup two external sip profiles with different external port each to handle traffic route to each sip provider. What is not clear is how handle multiple accounts with same sip provider? Can all the accounts share the same sip profile? Can Freeswitch register multiple accounts at once on the same sip profile? How is the dialplan gonna to work in this case? Examples to illustrate would be good.<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'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Cheers,<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Louie<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><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"'>From:</span></b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'> Steven Ayre [<a href="mailto:steveayre@gmail.com">mailto:steveayre@gmail.com</a>] <br><b>Sent:</b> Thursday, 24 July 2014 1:06 AM<br><b>To:</b> FreeSWITCH Users Help<br><b>Subject:</b> Re: [Freeswitch-users] multiple sip gateway configuration<o:p></o:p></span></p></div><p class=MsoNormal><o:p> </o:p></p><div><p class=MsoNormal>A single profile can send to any number of gateways.<o:p></o:p></p><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>You only need multiple profiles if you need to listen in multiple places, usually to handle either multiple IP addresses or different configurations on different ports.<o:p></o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>sip-ip and rtp-ip are the local IP of the server where you're receiving packets. It must be a real address, 0.0.0.0 (ie any address) isn't valid. If you're behind NAT they should be the internal IP, there are settings such as ext-sip-ip and ext-rtp-ip for handling NAT traversal.<o:p></o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>How to select the gateway to use will be very dependant on your use-case. However in general, you pick the destination gateway in the dialstring you bridge with, eg<o:p></o:p></p></div><div><p class=MsoNormal><action application="bridge" data="sofia/gateway/<span style='font-size:10.0pt;font-family:"Arial","sans-serif"'>Sip_Provider_A/12345"/></span><o:p></o:p></p></div><div><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Arial","sans-serif"'><action application="bridge" data="sofia/gateway/Sip_Provider_B/12345"/></span><o:p></o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>How you pick it is up to you, but as some examples you could set it as a variable in the user directory or based on conditions in the dialstring, you could use mod_distributor, mod_lcr etc, or try each in sequence as above.<o:p></o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>Steve<o:p></o:p></p></div></div><div><p class=MsoNormal style='margin-bottom:12.0pt'><o:p> </o:p></p><div><p class=MsoNormal>On 23 July 2014 14:48, Louie Liu <<a href="mailto:lliu@multinet.net.au" target="_blank">lliu@multinet.net.au</a>> wrote:<o:p></o:p></p><div><div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>Hi,<o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>There is requirement for me to add multiple sip gateways to Freeswitch, the first sip gateway uses the external.xml in sip profile to route traffic to Sip provider A. The second sip gateway will use a different UA to route traffic to Sip provider B. Here are my questions:<o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <o:p></o:p></p><p style='margin-left:38.25pt'>1.<span style='font-size:7.0pt'> </span>Do I need to create another external.xml with its own ip and port which point to the Sip Provider B? if that’s case, what do I need to change in the sip profile external.xml file? <o:p></o:p></p><p style='margin-left:38.25pt'>2.<span style='font-size:7.0pt'> </span>What is the rtp-ip and sip-ip in the external.xml file? Should that be the local IP of my sip server? <o:p></o:p></p><p style='margin-left:38.25pt'>3.<span style='font-size:7.0pt'> </span>What do I need to change in my dialplan so that I can reference the Sip_Provider_B? an example would be good.<o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>External.xml<o:p></o:p></p><div style='border:solid windowtext 1.0pt;padding:1.0pt 4.0pt 1.0pt 4.0pt'><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <param name="sip-port" value="$${external_sip_port}"/><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <param name="dialplan" value="XML"/><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <param name="context" value="public"/><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <param name="rtp-ip" value="$${local_ip_v4}"/><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <param name="sip-ip" value="$${local_ip_v4}"/><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <param name="ext-rtp-ip" value="1.2.3.4"/><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <param name="ext-sip-ip" value="1.2.3.4"/><o:p></o:p></p></div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>External/sip_provider_A.xml<o:p></o:p></p><div style='border:solid windowtext 1.0pt;padding:1.0pt 4.0pt 1.0pt 4.0pt'><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <gateway name="Sip_Provider_A"><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <param name="username" value="xxx"/><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <param name="password" value="xxx"/><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <param name="from-user" value="03xxxxxxx"/><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <param name="from-domain" value="x.y.z"/><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <param name="auth-username" value="yyyyy"/><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <param name="realm" value="<a href="http://bwas02.voip.izzz.zz.au" target="_blank">bwas02.voip.izzz.zz.au</a>"/><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <param name="proxy" value="<a href="http://sipconnect.voip.zzzz.zz.au" target="_blank">sipconnect.voip.zzzz.zz.au</a>"/><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <param name="register" value="true"/><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <param name="register-transport" value="udp"/><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <param name="context" value="public"/><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> </gateway> <o:p></o:p></p></div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>External/sip_provider_B.xml<o:p></o:p></p><div style='border:solid windowtext 1.0pt;padding:1.0pt 4.0pt 1.0pt 4.0pt'><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <gateway name="Sip_Provider_B"><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <param name="username" value="aaa"/><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <param name="password" value="bbb"/><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <param name="from-user" value="03xxxxxxx"/><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <param name="from-domain" value="x.y.z"/><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <param name="auth-username" value="yyyyy"/><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <param name="realm" value="<a href="http://bwas02.voip.izzz.zz.au" target="_blank">bwas02.voip.izzz.zz.au</a>"/><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <param name="proxy" value="<a href="http://sipconnect.voip.zzzz.zz.au" target="_blank">sipconnect.voip.zzzz.zz.au</a>"/><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <param name="register" value="true"/><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <param name="register-transport" value="udp"/><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <param name="context" value="public"/><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> </gateway> <o:p></o:p></p></div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>Cheers,<o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>Louie<o:p></o:p></p></div></div><p class=MsoNormal style='margin-bottom:12.0pt'><br>_________________________________________________________________________<br>Professional FreeSWITCH Consulting Services:<br><a href="mailto:consulting@freeswitch.org">consulting@freeswitch.org</a><br><a href="http://www.freeswitchsolutions.com" target="_blank">http://www.freeswitchsolutions.com</a><br><br>FreeSWITCH-powered IP PBX: The CudaTel Communication Server<br><a href="http://www.cudatel.com" target="_blank">http://www.cudatel.com</a><br><br>Official FreeSWITCH Sites<br><a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><br><a href="http://wiki.freeswitch.org" target="_blank">http://wiki.freeswitch.org</a><br><a href="http://www.cluecon.com" target="_blank">http://www.cluecon.com</a><br><br>FreeSWITCH-users mailing list<br><a href="mailto:FreeSWITCH-users@lists.freeswitch.org">FreeSWITCH-users@lists.freeswitch.org</a><br><a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a><br>UNSUBSCRIBE:<a href="http://lists.freeswitch.org/mailman/options/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/options/freeswitch-users</a><br><a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><o:p></o:p></p></div><p class=MsoNormal><o:p> </o:p></p></div></div></body></html>