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</o:shapelayout></xml><![endif]--></head><body lang=DE-CH link=blue vlink=purple><div class=WordSection1><p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Oh, just one last IMPORTANT thing I forgot to mention. Mind the presentation flags. If the “calling party number” (Asterisk </span><strong><span lang=EN-US>${CALLINGPRES})</span></strong><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'> is flagged as restricted it is ILLEGAL to transmit the number element to the next peer/subscriber/useragent serviced by Asterisk, unless it is in the domain of another licensed operator subject to regulatory provisions to know such data. If the entity running the Asterisk is not an operator but a retail customer and you are the carrier running Switch B, you must strip the restricted number element out in SS7, which is not quite easy because retail customers regularly don’t use SS7 but rather Q.931/PRI, where switch vendors are more likely to provide such retail restrictions. <o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>So long, Florian<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'> <o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US 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 lang=DE style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>Von:</span></b><span lang=DE style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'> asterisk-ss7-bounces@lists.digium.com [mailto:asterisk-ss7-bounces@lists.digium.com] <b>Im Auftrag von </b>florian@gruendler.net<br><b>Gesendet:</b> Montag, 22. August 2011 15:14<br><b>An:</b> asterisk-ss7@lists.digium.com<br><b>Betreff:</b> [asterisk-ss7] WG: ISUP call forwarding<o:p></o:p></span></p></div></div><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Mosbah,<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Apologies, I didn’t READ all of your post. So I am to change my answer as follows: Switch A becomes rather irrelevant here, except for the point of certifying you the calling party number to be network provided and verified (else, you don’t trust calls CLI for voicemail authentication and reverse charged services). Switch B must be configured right to send the redirection parameters to the ISUP powered Asterisk PBX. Let’s say it will. Now I am not in a position to KNOW the code of LibSS7 to tell you what ISUP information element is stored in which channel variable. I suggest you to dump ALL possible ones with a NoOp function in your incoming context. You can find the variables here under the paragraph “Predefined Channel Variables”<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><a href="http://www.voip-info.org/wiki/index.php?page=Asterisk+variables">http://www.voip-info.org/wiki/index.php?page=Asterisk+variables</a><o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Maybe one of the LibSS7 devs will jump in and refer to the right section of the manual.<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Regards, Florian<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-US 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 lang=EN-US style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>Von:</span></b><span lang=EN-US style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'> florian@gruendler.net [mailto:florian@gruendler.net] <br><b>Gesendet:</b> Montag, 22. August 2011 14:51<br><b>An:</b> </span><span lang=DE style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>asterisk-ss7@lists.digium.com<br><b>Betreff:</b> AW: [asterisk-ss7] ISUP call forwarding<o:p></o:p></span></p></div></div><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Mosbah,<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>We are talking SS7/ISUP here right? What Switch (make,model,software) handles Subscriber B? ISUP Switch A will let ISUP Switch B know (if it was configured by people with a notion of the matter) the redirected number plus the ISUP redirection parameters but it is completely unknown in your equation how Switch B and Asterisk relate to eachother. Is Asterisk=Subscriber B or is there any gateway involved which follows any standard (e.g. ISUP<->SIP)?<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoPlainText><span lang=EN-US><a href="http://www.dialogic.com/webhelp/IMG1010/10.5.1/WebHelp/Description/SIP/sip_diversion_scen.htm">http://www.dialogic.com/webhelp/IMG1010/10.5.1/WebHelp/Description/SIP/sip_diversion_scen.htm</a><o:p></o:p></span></p><p class=MsoPlainText><span lang=EN-US><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>regards, Florian<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><div style='border:none;border-left:solid blue 1.5pt;padding:0cm 0cm 0cm 4.0pt'><div><div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0cm 0cm 0cm'><p class=MsoNormal><b><span lang=DE style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>Von:</span></b><span lang=DE style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'> asterisk-ss7-bounces@lists.digium.com [mailto:asterisk-ss7-bounces@lists.digium.com] <b>Im Auftrag von </b>mosbah abdelkader<br><b>Gesendet:</b> Montag, 22. August 2011 14:28<br><b>An:</b> asterisk-ss7@lists.digium.com<br><b>Betreff:</b> [asterisk-ss7] ISUP call forwarding<o:p></o:p></span></p></div></div><p class=MsoNormal><o:p> </o:p></p><div><p class=MsoNormal>Hi,<br><br>please can anyone help me on that situation:<br><br><br>Subsc. A ------------------------------Subsc. B --------------------------Asterisk<br><br><br>Subsc A calls B.<br><br>B forwards the call to Asterisk using ISUP.<br><br>Is the Subsc A number included in the parameters of the ISUP call forwarded from B to Asterisk?<br><br>Please help.<o:p></o:p></p></div></div></div></body></html>