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

Mon Jul 1 14:10:07 CDT 2019

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.

From: asterisk-users [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Israel Gottlieb
Sent: Monday, July 1, 2019 1:57 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com>
Subject: Re: [asterisk-users] asterisk-users Digest, Vol 179, Issue 1

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 <mailto: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 <mailto:jcolp at digium.com> >
To: asterisk-users at lists.digium.com <mailto: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 <mailto: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 <http://www.digium.com>  & www.asterisk.org <http://www.asterisk.org> 

------------------------------

Message: 2
Date: Mon, 1 Jul 2019 14:53:47 +0000
From: "Jason N" <support at telium.io <mailto:support at telium.io> >
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
        <asterisk-users at lists.digium.com <mailto: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 <mailto: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 <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 <mailto: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 <http://www.digium.com>  & www.asterisk.org <http://www.asterisk.org> 

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------------------------------

Message: 3
Date: Mon, 01 Jul 2019 11:57:45 -0300
From: "Joshua C. Colp" <jcolp at digium.com <mailto:jcolp at digium.com> >
To: asterisk-users at lists.digium.com <mailto: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 <mailto: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 <http://www.digium.com>  & www.asterisk.org <http://www.asterisk.org> 

------------------------------

Subject: Digest Footer

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      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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