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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'>We are not allowed to insert anything into the call path. So somehow we have get S included into call without adding anything into the call path. That’s why I thought a SIP JOIN would work (where device C would handle the multiparty call) – but it sounds like Asterisk doesn’t support that.<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><b><span style='font-size:11.0pt;font-family:"Calibri",sans-serif'>From:</span></b><span style='font-size:11.0pt;font-family:"Calibri",sans-serif'> asterisk-users [mailto:asterisk-users-bounces@lists.digium.com] <b>On Behalf Of </b>Israel Gottlieb<br><b>Sent:</b> Monday, July 1, 2019 1:57 PM<br><b>To:</b> Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com><br><b>Subject:</b> Re: [asterisk-users] asterisk-users Digest, Vol 179, Issue 1<o:p></o:p></span></p><p class=MsoNormal><o:p> </o:p></p><div><p class=MsoNormal>how about sticking in a pbx between [c] and [h]<o:p></o:p></p><div><p class=MsoNormal>so when [h] hangsup you send to [s] if that is 3rd party else i dont see how you could redirect [c] at all <o:p></o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>else maybe ask them to have [h] redirect [c] to [s] then [h] will also be out of the call<o:p></o:p></p></div></div><p class=MsoNormal><o:p> </o:p></p><div><div><p class=MsoNormal>On Mon, Jul 1, 2019, 20:03 <<a href="mailto:asterisk-users-request@lists.digium.com">asterisk-users-request@lists.digium.com</a> wrote:<o:p></o:p></p></div><blockquote style='border:none;border-left:solid #CCCCCC 1.0pt;padding:0in 0in 0in 6.0pt;margin-left:4.8pt;margin-right:0in'><p class=MsoNormal>Send asterisk-users mailing list submissions to<br> <a href="mailto:asterisk-users@lists.digium.com" target="_blank">asterisk-users@lists.digium.com</a><br><br>To subscribe or unsubscribe via the World Wide Web, visit<br> <a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br>or, via email, send a message with subject or body 'help' to<br> <a href="mailto:asterisk-users-request@lists.digium.com" target="_blank">asterisk-users-request@lists.digium.com</a><br><br>You can reach the person managing the list at<br> <a href="mailto:asterisk-users-owner@lists.digium.com" target="_blank">asterisk-users-owner@lists.digium.com</a><br><br>When replying, please edit your Subject line so it is more specific<br>than "Re: Contents of asterisk-users digest..."<br><br><br>Today's Topics:<br><br> 1. Re: Second Asterisk server SIP JOIN a call to conduct a<br> post-call survey (Joshua C. Colp)<br> 2. Re: Second Asterisk server SIP JOIN a call to conduct a<br> post-call survey (Jason N)<br> 3. Re: Second Asterisk server SIP JOIN a call to conduct a<br> post-call survey (Joshua C. Colp)<br><br><br>----------------------------------------------------------------------<br><br>Message: 1<br>Date: Mon, 01 Jul 2019 11:15:01 -0300<br>From: "Joshua C. Colp" <<a href="mailto:jcolp@digium.com" target="_blank">jcolp@digium.com</a>><br>To: <a href="mailto:asterisk-users@lists.digium.com" target="_blank">asterisk-users@lists.digium.com</a><br>Subject: Re: [asterisk-users] Second Asterisk server SIP JOIN a call<br> to conduct a post-call survey<br>Message-ID: <<a href="mailto:be3a1911-7870-4039-9a35-39f7b5be81c4@www.fastmail.com" target="_blank">be3a1911-7870-4039-9a35-39f7b5be81c4@www.fastmail.com</a>><br>Content-Type: text/plain;charset=utf-8<br><br>On Sun, Jun 30, 2019, at 11:09 AM, Jason N wrote:<br>> I am designing a solution for a hotel booking call center with the <br>> following (mandatory) design: After the call from the customer with the <br>> booking agent is complete (and the Hotel PBX disconnects from the <br>> call), a second PBX takes over to conduct a survey of how the call <br>> went. Both PBX’s are Asterisk based. <br>> <br>> <br>> So customer phone [C] connects to hotel PBX [H]. Once [H] disconnects, <br>> the survey PBX [S] grabs the call and conducts the survey. [H] must <br>> completely disconnect from the call before [S] can start the survey. <br>> [H] cannot transfer/forward the call to [S]. <br>> <br>> <br>> At a high level the solution seems to be: On [C] connection to [H], [H] <br>> sends call information to [S]. [S] issues a SIP JOIN to [C] and joins <br>> the call. [S] somehow detects that [H] has disconnected and then begins <br>> the survey.<br>> <br>> <br>> Would the above work conceptually? If so, how do I tell Asterisk [S] to <br>> contact [C] and join the call already in progress? (I can get call info <br>> from [H] to [S]).<br><br>It would be easiest for H to just Dial S after the first call leg is done. This can be done using the 'g' option to Dial[1] which continues dialplan application after the outgoing call leg hangs up. You could even send information as SIP headers if need be so S sees the info.<br><br>[1] <a href="https://wiki.asterisk.org/wiki/display/AST/Asterisk+16+Application_Dial" target="_blank">https://wiki.asterisk.org/wiki/display/AST/Asterisk+16+Application_Dial</a><br><br>-- <br>Joshua C. Colp<br>Digium - A Sangoma Company | Senior Software Developer<br>445 Jan Davis Drive NW - Huntsville, AL 35806 - US<br>Check us out at: <a href="http://www.digium.com" target="_blank">www.digium.com</a> & <a href="http://www.asterisk.org" target="_blank">www.asterisk.org</a><br><br><br><br>------------------------------<br><br>Message: 2<br>Date: Mon, 1 Jul 2019 14:53:47 +0000<br>From: "Jason N" <<a href="mailto:support@telium.io" target="_blank">support@telium.io</a>><br>To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"<br> <<a href="mailto:asterisk-users@lists.digium.com" target="_blank">asterisk-users@lists.digium.com</a>><br>Subject: Re: [asterisk-users] Second Asterisk server SIP JOIN a call<br> to conduct a post-call survey<br>Message-ID:<br> <<a href="mailto:0100016bae071017-8cd5329f-5e33-493c-a339-c997586e4708-000000@email.amazonses.com" target="_blank">0100016bae071017-8cd5329f-5e33-493c-a339-c997586e4708-000000@email.amazonses.com</a>><br><br>Content-Type: text/plain; charset="utf-8"<br><br>Unfortunately I am not allowed any changes to H's PBX / dialplan. The restriction I have is that upon H's total disconnection from C, that S continues the call with C. That's why I thought that if I could get S to SIP JOIN the call from C, that once H disconnects S can continue. I can extract the SIP call info on H and pass that to S (so it can join the call). <br><br>I'm just not sure if this concept is possible/practical.<br><br><br>-----Original Message-----<br>From: asterisk-users [mailto:<a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank">asterisk-users-bounces@lists.digium.com</a>] On Behalf Of Joshua C. Colp<br>Sent: Monday, July 1, 2019 10:15 AM<br>To: <a href="mailto:asterisk-users@lists.digium.com" target="_blank">asterisk-users@lists.digium.com</a><br>Subject: Re: [asterisk-users] Second Asterisk server SIP JOIN a call to conduct a post-call survey<br><br>On Sun, Jun 30, 2019, at 11:09 AM, Jason N wrote:<br>> I am designing a solution for a hotel booking call center with the <br>> following (mandatory) design: After the call from the customer with <br>> the booking agent is complete (and the Hotel PBX disconnects from the <br>> call), a second PBX takes over to conduct a survey of how the call <br>> went. Both PBX’s are Asterisk based.<br>> <br>> <br>> So customer phone [C] connects to hotel PBX [H]. Once [H] disconnects, <br>> the survey PBX [S] grabs the call and conducts the survey. [H] must <br>> completely disconnect from the call before [S] can start the survey.<br>> [H] cannot transfer/forward the call to [S]. <br>> <br>> <br>> At a high level the solution seems to be: On [C] connection to [H], <br>> [H] sends call information to [S]. [S] issues a SIP JOIN to [C] and <br>> joins the call. [S] somehow detects that [H] has disconnected and then <br>> begins the survey.<br>> <br>> <br>> Would the above work conceptually? If so, how do I tell Asterisk [S] <br>> to contact [C] and join the call already in progress? (I can get call <br>> info from [H] to [S]).<br><br>It would be easiest for H to just Dial S after the first call leg is done. This can be done using the 'g' option to Dial[1] which continues dialplan application after the outgoing call leg hangs up. You could even send information as SIP headers if need be so S sees the info.<br><br>[1] <a href="https://wiki.asterisk.org/wiki/display/AST/Asterisk+16+Application_Dial" target="_blank">https://wiki.asterisk.org/wiki/display/AST/Asterisk+16+Application_Dial</a><br><br>--<br>Joshua C. Colp<br>Digium - A Sangoma Company | Senior Software Developer<br>445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: <a href="http://www.digium.com" target="_blank">www.digium.com</a> & <a href="http://www.asterisk.org" target="_blank">www.asterisk.org</a><br><br>--<br>_____________________________________________________________________<br>-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br><br>Check out the new Asterisk community forum at: <a href="https://community.asterisk.org/" target="_blank">https://community.asterisk.org/</a><br><br>New to Asterisk? Start here:<br> <a href="https://wiki.asterisk.org/wiki/display/AST/Getting+Started" target="_blank">https://wiki.asterisk.org/wiki/display/AST/Getting+Started</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><br><br><br><br><br>------------------------------<br><br>Message: 3<br>Date: Mon, 01 Jul 2019 11:57:45 -0300<br>From: "Joshua C. Colp" <<a href="mailto:jcolp@digium.com" target="_blank">jcolp@digium.com</a>><br>To: <a href="mailto:asterisk-users@lists.digium.com" target="_blank">asterisk-users@lists.digium.com</a><br>Subject: Re: [asterisk-users] Second Asterisk server SIP JOIN a call<br> to conduct a post-call survey<br>Message-ID: <<a href="mailto:27f60411-06d6-4f75-a356-ca151e1f2505@www.fastmail.com" target="_blank">27f60411-06d6-4f75-a356-ca151e1f2505@www.fastmail.com</a>><br>Content-Type: text/plain<br><br>On Mon, Jul 1, 2019, at 11:54 AM, Jason N wrote:<br>> Unfortunately I am not allowed any changes to H's PBX / dialplan. <br>> The restriction I have is that upon H's total disconnection from C, <br>> that S continues the call with C. That's why I thought that if I could <br>> get S to SIP JOIN the call from C, that once H disconnects S can <br>> continue. I can extract the SIP call info on H and pass that to S (so <br>> it can join the call). <br>> <br>> I'm just not sure if this concept is possible/practical.<br><br>There is no such thing as "joining" a call like that in Asterisk. It would be trying to do server side three way calling, which is not supported like that.<br><br>-- <br>Joshua C. Colp<br>Digium - A Sangoma Company | Senior Software Developer<br>445 Jan Davis Drive NW - Huntsville, AL 35806 - US<br>Check us out at: <a href="http://www.digium.com" target="_blank">www.digium.com</a> & <a href="http://www.asterisk.org" target="_blank">www.asterisk.org</a><br><br><br><br>------------------------------<br><br>Subject: Digest Footer<br><br>_______________________________________________<br>--Bandwidth and Colocation Provided by <a href="http://www.api-digital.com--" target="_blank">http://www.api-digital.com--</a><br><br>Check out the new Asterisk community forum at: <a href="https://community.asterisk.org/" target="_blank">https://community.asterisk.org/</a><br><br>New to Asterisk? Start here:<br> <a href="https://wiki.asterisk.org/wiki/display/AST/Getting+Started" target="_blank">https://wiki.asterisk.org/wiki/display/AST/Getting+Started</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><br><br>------------------------------<br><br>End of asterisk-users Digest, Vol 179, Issue 1<br>**********************************************<o:p></o:p></p></blockquote></div></div></body></html>