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</o:shapelayout></xml><![endif]--></head><body lang=HU link=blue vlink=purple><div class=WordSection1><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri",sans-serif;color:#1F497D;mso-fareast-language:EN-US'>Hi,<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri",sans-serif;color:#1F497D;mso-fareast-language:EN-US'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri",sans-serif;color:#1F497D;mso-fareast-language:EN-US'>Unfortunately this is not the case.<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri",sans-serif;color:#1F497D;mso-fareast-language:EN-US'>For some reasons (separation, fraud detection) they separate the customer using their source IP address. So I am still looking for some solution.<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri",sans-serif;color:#1F497D;mso-fareast-language:EN-US'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri",sans-serif;color:#1F497D;mso-fareast-language:EN-US'>Thanks,<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri",sans-serif;color:#1F497D;mso-fareast-language:EN-US'>Attila<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri",sans-serif;color:#1F497D;mso-fareast-language:EN-US'><o:p> </o:p></span></p><p class=MsoNormal><b><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri",sans-serif'>From:</span></b><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri",sans-serif'> asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] <b>On Behalf Of </b>Glenn Geller (VDOPh)<br><b>Sent:</b> Wednesday, May 25, 2016 11:47 PM<br><b>To:</b> Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com><br><b>Subject:</b> Re: [asterisk-users] Sending Calls via SIP trunk from several different IP addresses from same Asterisk Machine, to the same destination IP<o:p></o:p></span></p><p class=MsoNormal><o:p> </o:p></p><div><p class=MsoNormal>Hi,<o:p></o:p></p><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>Usually, the trunk provider(s) provide a mechanism to support this, and it's the "Tech Prefix" or just "Prefix".<o:p></o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>So, for Tenant 1, send 12348005551212 and for Tenant 2, send 56788005551212.<o:p></o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>They'll then strip the prefix, and send along to 8005551212<o:p></o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>Most trunk providers worth anything will support this type of termination.<o:p></o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>Hope this helps,<o:p></o:p></p></div><div><p class=MsoNormal><br clear=all><o:p></o:p></p><div><div><p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Arial",sans-serif;color:#222222'>Glenn @ VDO</span></b><o:p></o:p></p></div></div><p class=MsoNormal><o:p> </o:p></p><div><p class=MsoNormal>On Wed, May 25, 2016 at 2:13 PM, Attila Megyeri <<a href="mailto:amegyeri@minerva-soft.com" target="_blank">amegyeri@minerva-soft.com</a>> wrote:<o:p></o:p></p><blockquote style='border:none;border-left:solid #CCCCCC 1.0pt;padding:0cm 0cm 0cm 6.0pt;margin-left:4.8pt;margin-right:0cm'><div><div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US>Hi!</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US> </span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US>I would like to reopen a discussion that I saw a couple of years ago, with the subject “Sending Calls via SIP trunk from two different IP addresses from same Asterisk Machine”</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US> </span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US>The use case is simpe: There are providers that want to see a separate source IP address for each of their customers SIP trunks. Now, if we have an asterisk box with several customers, we have a problem.</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US> </span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US>Does anyone have experience in this topic? How could we send outgoing calls (to the same destination IP) from different source IPs depending on the caller ID (Based on From: field, sip account, preferred-identity, whatever).</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US> </span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US>I was thinking about some Kamailio, or SBC that would take the calls from asterisk using a user/pass authentication on a single interface, and initiate calls from a dedicated IP address for each customer.</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US>Any better idea?</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US> </span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US>Thanks</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US style='color:#888888'> </span><span style='color:#888888'><o:p></o:p></span></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span lang=EN-US style='color:#888888'>Attila</span><span style='color:#888888'><o:p></o:p></span></p></div></div><p class=MsoNormal><br>--<br>_____________________________________________________________________<br>-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>New to Asterisk? Join us for a live introductory webinar every Thurs:<br> <a href="http://www.asterisk.org/hello" target="_blank">http://www.asterisk.org/hello</a><br><br>asterisk-users mailing list<br>To UNSUBSCRIBE or update options visit:<br> <a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><o:p></o:p></p></blockquote></div><p class=MsoNormal><o:p> </o:p></p></div></div></div></body></html>