Re: [asterisk-users] Second Asterisk server SIP JOIN a call to conduct a post-call survey
On Mon, Jul 1, 2019, at 11:54 AM, Jason N wrote: > 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). > > I'm just not sure if this concept is possible/practical. 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. -- Joshua C. Colp Digium - A Sangoma Company | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Second Asterisk server SIP JOIN a call to conduct a post-call survey
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). I'm just not sure if this concept is possible/practical. -Original Message- From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joshua C. Colp Sent: Monday, July 1, 2019 10:15 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Second Asterisk server SIP JOIN a call to conduct a post-call survey On Sun, Jun 30, 2019, at 11:09 AM, Jason N wrote: > I am designing a solution for a hotel booking call center with the > following (mandatory) design: After the call from the customer with > the booking agent is complete (and the Hotel PBX disconnects from the > call), a second PBX takes over to conduct a survey of how the call > went. Both PBX’s are Asterisk based. > > > So customer phone [C] connects to hotel PBX [H]. Once [H] disconnects, > the survey PBX [S] grabs the call and conducts the survey. [H] must > completely disconnect from the call before [S] can start the survey. > [H] cannot transfer/forward the call to [S]. > > > At a high level the solution seems to be: On [C] connection to [H], > [H] sends call information to [S]. [S] issues a SIP JOIN to [C] and > joins the call. [S] somehow detects that [H] has disconnected and then > begins the survey. > > > Would the above work conceptually? If so, how do I tell Asterisk [S] > to contact [C] and join the call already in progress? (I can get call > info from [H] to [S]). 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. [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+16+Application_Dial -- Joshua C. Colp Digium - A Sangoma Company | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Second Asterisk server SIP JOIN a call to conduct a post-call survey
On Sun, Jun 30, 2019, at 11:09 AM, Jason N wrote: > I am designing a solution for a hotel booking call center with the > following (mandatory) design: After the call from the customer with the > booking agent is complete (and the Hotel PBX disconnects from the > call), a second PBX takes over to conduct a survey of how the call > went. Both PBX’s are Asterisk based. > > > So customer phone [C] connects to hotel PBX [H]. Once [H] disconnects, > the survey PBX [S] grabs the call and conducts the survey. [H] must > completely disconnect from the call before [S] can start the survey. > [H] cannot transfer/forward the call to [S]. > > > At a high level the solution seems to be: On [C] connection to [H], [H] > sends call information to [S]. [S] issues a SIP JOIN to [C] and joins > the call. [S] somehow detects that [H] has disconnected and then begins > the survey. > > > Would the above work conceptually? If so, how do I tell Asterisk [S] to > contact [C] and join the call already in progress? (I can get call info > from [H] to [S]). 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. [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+16+Application_Dial -- Joshua C. Colp Digium - A Sangoma Company | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Second Asterisk server SIP JOIN a call to conduct a post-call survey
And, how would [S] know that [H] has disconnected? (Is there an Asterisk event that indicates one party has disconnected from a multi-party call) From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jason N Sent: Sunday, June 30, 2019 10:08 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Second Asterisk server SIP JOIN a call to conduct a post-call survey I am designing a solution for a hotel booking call center with the following (mandatory) design: After the call from the customer with the booking agent is complete (and the Hotel PBX disconnects from the call), a second PBX takes over to conduct a survey of how the call went. Both PBX's are Asterisk based. So customer phone [C] connects to hotel PBX [H]. Once [H] disconnects, the survey PBX [S] grabs the call and conducts the survey. [H] must completely disconnect from the call before [S] can start the survey. [H] cannot transfer/forward the call to [S]. At a high level the solution seems to be: On [C] connection to [H], [H] sends call information to [S]. [S] issues a SIP JOIN to [C] and joins the call. [S] somehow detects that [H] has disconnected and then begins the survey. Would the above work conceptually? If so, how do I tell Asterisk [S] to contact [C] and join the call already in progress? (I can get call info from [H] to [S]). Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Second Asterisk server SIP JOIN a call to conduct a post-call survey
I am designing a solution for a hotel booking call center with the following (mandatory) design: After the call from the customer with the booking agent is complete (and the Hotel PBX disconnects from the call), a second PBX takes over to conduct a survey of how the call went. Both PBX's are Asterisk based. So customer phone [C] connects to hotel PBX [H]. Once [H] disconnects, the survey PBX [S] grabs the call and conducts the survey. [H] must completely disconnect from the call before [S] can start the survey. [H] cannot transfer/forward the call to [S]. At a high level the solution seems to be: On [C] connection to [H], [H] sends call information to [S]. [S] issues a SIP JOIN to [C] and joins the call. [S] somehow detects that [H] has disconnected and then begins the survey. Would the above work conceptually? If so, how do I tell Asterisk [S] to contact [C] and join the call already in progress? (I can get call info from [H] to [S]). Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
