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

2019-07-01 Thread Jason N 
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-boun...@lists.digium.com] On Behalf 
Of Israel Gottlieb
Sent: Monday, July 1, 2019 1:57 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion 

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 mailto:asterisk-users-requ...@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" mailto:jc...@digium.com> >
To: asterisk-users@lists.digium.com  
Subject: Re: [asterisk-users] Second Asterisk server SIP JOIN a call
to conduct  a post-call survey
Message-ID: mailto:be3a1911-7870-4039-9a35-39f7b5be8...@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 +
From: "Jason N" mailto:supp...@telium.io> >
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
mailto:asterisk-users@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-c997586e4708-000...@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

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

2019-07-01 Thread Israel Gottlieb 
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  Send asterisk-users mailing list submissions to
> asterisk-users@lists.digium.com
>
> To subscribe or unsubscribe via the World Wide Web, visit
> http://lists.digium.com/mailman/listinfo/asterisk-users
> or, via email, send a message with subject or body 'help' to
> asterisk-users-requ...@lists.digium.com
>
> You can reach the person managing the list at
> asterisk-users-ow...@lists.digium.com
>
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of asterisk-users digest..."
>
>
> 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" 
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] Second Asterisk server SIP JOIN a call
> to conduct  a post-call survey
> Message-ID: 
> 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 +
> From: "Jason N" 
> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
> 
> Subject: Re: [asterisk-users] Second Asterisk server SIP JOIN a call
> to  conduct a post-call survey
> Message-ID:
> <
> 0100016bae071017-8cd5329f-5e33-493c-a339-c997586e4708-000...@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-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

Re: [asterisk-users] Second Asterisk server SIP JOIN a call to conduct a post-call survey

2019-07-01 Thread Joshua C. Colp 
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

-- 
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Re: [asterisk-users] Second Asterisk server SIP JOIN a call to conduct a post-call survey

2019-07-01 Thread Jason N 
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

--
_
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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
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Re: [asterisk-users] Second Asterisk server SIP JOIN a call to conduct a post-call survey

2019-07-01 Thread Joshua C. Colp 
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