[asterisk-users] asterisk-users Digest, Vol 179, Issue 1

Mon Jul 1 12:57:07 CDT 2019

how about sticking in a pbx between [c] and [h]
so when [h] hangsup you send to [s] if that is 3rd party else i dont see
how you could redirect [c] at all

else maybe ask them to have [h] redirect [c] to [s] then [h] will also be
out of the call

On Mon, Jul 1, 2019, 20:03 <asterisk-users-request at lists.digium.com wrote:

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> Today's Topics:
>
>    1. Re: Second Asterisk server SIP JOIN a call to conduct     a
>       post-call survey (Joshua C. Colp)
>    2. Re: Second Asterisk server SIP JOIN a call to     conduct a
>       post-call survey (Jason N)
>    3. Re: Second Asterisk server SIP JOIN a call to     conduct a
>       post-call survey (Joshua C. Colp)
>
>
> ----------------------------------------------------------------------
>
> Message: 1
> Date: Mon, 01 Jul 2019 11:15:01 -0300
> From: "Joshua C. Colp" <jcolp at digium.com>
> To: asterisk-users at lists.digium.com
> Subject: Re: [asterisk-users] Second Asterisk server SIP JOIN a call
>         to conduct      a post-call survey
> Message-ID: <be3a1911-7870-4039-9a35-39f7b5be81c4 at www.fastmail.com>
> Content-Type: text/plain;charset=utf-8
>
> 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
>
>
>
> ------------------------------
>
> Message: 2
> Date: Mon, 1 Jul 2019 14:53:47 +0000
> From: "Jason N" <support at telium.io>
> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
>         <asterisk-users at lists.digium.com>
> Subject: Re: [asterisk-users] Second Asterisk server SIP JOIN a call
>         to      conduct a post-call survey
> Message-ID:
>         <
> 0100016bae071017-8cd5329f-5e33-493c-a339-c997586e4708-000000 at email.amazonses.com
> >
>
> Content-Type: text/plain;       charset="utf-8"
>
> 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 at lists.digium.com] On
> Behalf Of Joshua C. Colp
> Sent: Monday, July 1, 2019 10:15 AM
> To: asterisk-users at 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
>
> --
> _____________________________________________________________________
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>
> Check out the new Asterisk community forum at:
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>
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>       https://wiki.asterisk.org/wiki/display/AST/Getting+Started
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>
> ------------------------------
>
> Message: 3
> Date: Mon, 01 Jul 2019 11:57:45 -0300
> From: "Joshua C. Colp" <jcolp at digium.com>
> To: asterisk-users at lists.digium.com
> Subject: Re: [asterisk-users] Second Asterisk server SIP JOIN a call
>         to      conduct a post-call survey
> Message-ID: <27f60411-06d6-4f75-a356-ca151e1f2505 at www.fastmail.com>
> Content-Type: text/plain
>
> 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
>
>
>
> ------------------------------
>
> Subject: Digest Footer
>
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> Check out the new Asterisk community forum at:
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>
> New to Asterisk? Start here:
>       https://wiki.asterisk.org/wiki/display/AST/Getting+Started
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> ------------------------------
>
> End of asterisk-users Digest, Vol 179, Issue 1
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