asterisk users - Jun 2019 - 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

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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 at lists.digium.com] On
Behalf Of Jason N
Sent: Sunday, June 30, 2019 10:08 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
<asterisk-users at lists.digium.com>
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

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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

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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