Second Asterisk Server SIP JOIN A Call To Conduct A Post-call Survey

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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

2 thoughts on - 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-bounces@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

  • 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-bounces@lists.digium.com] 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

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    445 Jan Davis Drive NW – Huntsville, AL 35806 – US Check us out at: http://www.digium.com & http://www.asterisk.org

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