Israel Gottlieb
2019-Jul-01 17:57 UTC
[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 wrote:> Send asterisk-users mailing list submissions to > asterisk-users at 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-request at lists.digium.com > > You can reach the person managing the list at > asterisk-users-owner at 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" <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 > > -- > _____________________________________________________________________ > -- 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 > > > > > ------------------------------ > > 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 > > _______________________________________________ > --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 > > ------------------------------ > > End of asterisk-users Digest, Vol 179, Issue 1 > ********************************************** >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20190701/56567c4f/attachment.html>
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:
Send asterisk-users mailing list submissions to
asterisk-users at lists.digium.com <mailto:asterisk-users at
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-request at lists.digium.com
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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" <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>
--
_____________________________________________________________________
-- 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
------------------------------
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
_______________________________________________
--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
------------------------------
End of asterisk-users Digest, Vol 179, Issue 1
**********************************************
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