<div dir="auto">how about sticking in a pbx between [c] and [h]<div dir="auto">so when [h] hangsup you send to [s] if that is 3rd party else i dont see how you could redirect [c] at all </div><div dir="auto"><br></div><div dir="auto">else maybe ask them to have [h] redirect [c] to [s] then [h] will also be out of the call</div></div><br><div class="gmail_quote"><div dir="ltr">On Mon, Jul 1, 2019, 20:03 <<a href="mailto:asterisk-users-request@lists.digium.com">asterisk-users-request@lists.digium.com</a> wrote:<br></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Send asterisk-users mailing list submissions to<br>
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Today's Topics:<br>
<br>
1. Re: Second Asterisk server SIP JOIN a call to conduct a<br>
post-call survey (Joshua C. Colp)<br>
2. Re: Second Asterisk server SIP JOIN a call to conduct a<br>
post-call survey (Jason N)<br>
3. Re: Second Asterisk server SIP JOIN a call to conduct a<br>
post-call survey (Joshua C. Colp)<br>
<br>
<br>
----------------------------------------------------------------------<br>
<br>
Message: 1<br>
Date: Mon, 01 Jul 2019 11:15:01 -0300<br>
From: "Joshua C. Colp" <<a href="mailto:jcolp@digium.com" target="_blank" rel="noreferrer">jcolp@digium.com</a>><br>
To: <a href="mailto:asterisk-users@lists.digium.com" target="_blank" rel="noreferrer">asterisk-users@lists.digium.com</a><br>
Subject: Re: [asterisk-users] Second Asterisk server SIP JOIN a call<br>
to conduct a post-call survey<br>
Message-ID: <<a href="mailto:be3a1911-7870-4039-9a35-39f7b5be81c4@www.fastmail.com" target="_blank" rel="noreferrer">be3a1911-7870-4039-9a35-39f7b5be81c4@www.fastmail.com</a>><br>
Content-Type: text/plain;charset=utf-8<br>
<br>
On Sun, Jun 30, 2019, at 11:09 AM, Jason N wrote:<br>
> I am designing a solution for a hotel booking call center with the <br>
> following (mandatory) design: After the call from the customer with the <br>
> booking agent is complete (and the Hotel PBX disconnects from the <br>
> call), a second PBX takes over to conduct a survey of how the call <br>
> went. Both PBX’s are Asterisk based. <br>
> <br>
> <br>
> So customer phone [C] connects to hotel PBX [H]. Once [H] disconnects, <br>
> the survey PBX [S] grabs the call and conducts the survey. [H] must <br>
> completely disconnect from the call before [S] can start the survey. <br>
> [H] cannot transfer/forward the call to [S]. <br>
> <br>
> <br>
> At a high level the solution seems to be: On [C] connection to [H], [H] <br>
> sends call information to [S]. [S] issues a SIP JOIN to [C] and joins <br>
> the call. [S] somehow detects that [H] has disconnected and then begins <br>
> the survey.<br>
> <br>
> <br>
> Would the above work conceptually? If so, how do I tell Asterisk [S] to <br>
> contact [C] and join the call already in progress? (I can get call info <br>
> from [H] to [S]).<br>
<br>
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.<br>
<br>
[1] <a href="https://wiki.asterisk.org/wiki/display/AST/Asterisk+16+Application_Dial" rel="noreferrer noreferrer" target="_blank">https://wiki.asterisk.org/wiki/display/AST/Asterisk+16+Application_Dial</a><br>
<br>
-- <br>
Joshua C. Colp<br>
Digium - A Sangoma Company | Senior Software Developer<br>
445 Jan Davis Drive NW - Huntsville, AL 35806 - US<br>
Check us out at: <a href="http://www.digium.com" rel="noreferrer noreferrer" target="_blank">www.digium.com</a> & <a href="http://www.asterisk.org" rel="noreferrer noreferrer" target="_blank">www.asterisk.org</a><br>
<br>
<br>
<br>
------------------------------<br>
<br>
Message: 2<br>
Date: Mon, 1 Jul 2019 14:53:47 +0000<br>
From: "Jason N" <<a href="mailto:support@telium.io" target="_blank" rel="noreferrer">support@telium.io</a>><br>
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"<br>
<<a href="mailto:asterisk-users@lists.digium.com" target="_blank" rel="noreferrer">asterisk-users@lists.digium.com</a>><br>
Subject: Re: [asterisk-users] Second Asterisk server SIP JOIN a call<br>
to conduct a post-call survey<br>
Message-ID:<br>
<<a href="mailto:0100016bae071017-8cd5329f-5e33-493c-a339-c997586e4708-000000@email.amazonses.com" target="_blank" rel="noreferrer">0100016bae071017-8cd5329f-5e33-493c-a339-c997586e4708-000000@email.amazonses.com</a>><br>
<br>
Content-Type: text/plain; charset="utf-8"<br>
<br>
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). <br>
<br>
I'm just not sure if this concept is possible/practical.<br>
<br>
<br>
-----Original Message-----<br>
From: asterisk-users [mailto:<a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank" rel="noreferrer">asterisk-users-bounces@lists.digium.com</a>] On Behalf Of Joshua C. Colp<br>
Sent: Monday, July 1, 2019 10:15 AM<br>
To: <a href="mailto:asterisk-users@lists.digium.com" target="_blank" rel="noreferrer">asterisk-users@lists.digium.com</a><br>
Subject: Re: [asterisk-users] Second Asterisk server SIP JOIN a call to conduct a post-call survey<br>
<br>
On Sun, Jun 30, 2019, at 11:09 AM, Jason N wrote:<br>
> I am designing a solution for a hotel booking call center with the <br>
> following (mandatory) design: After the call from the customer with <br>
> the booking agent is complete (and the Hotel PBX disconnects from the <br>
> call), a second PBX takes over to conduct a survey of how the call <br>
> went. Both PBX’s are Asterisk based.<br>
> <br>
> <br>
> So customer phone [C] connects to hotel PBX [H]. Once [H] disconnects, <br>
> the survey PBX [S] grabs the call and conducts the survey. [H] must <br>
> completely disconnect from the call before [S] can start the survey.<br>
> [H] cannot transfer/forward the call to [S]. <br>
> <br>
> <br>
> At a high level the solution seems to be: On [C] connection to [H], <br>
> [H] sends call information to [S]. [S] issues a SIP JOIN to [C] and <br>
> joins the call. [S] somehow detects that [H] has disconnected and then <br>
> begins the survey.<br>
> <br>
> <br>
> Would the above work conceptually? If so, how do I tell Asterisk [S] <br>
> to contact [C] and join the call already in progress? (I can get call <br>
> info from [H] to [S]).<br>
<br>
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.<br>
<br>
[1] <a href="https://wiki.asterisk.org/wiki/display/AST/Asterisk+16+Application_Dial" rel="noreferrer noreferrer" target="_blank">https://wiki.asterisk.org/wiki/display/AST/Asterisk+16+Application_Dial</a><br>
<br>
--<br>
Joshua C. Colp<br>
Digium - A Sangoma Company | Senior Software Developer<br>
445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: <a href="http://www.digium.com" rel="noreferrer noreferrer" target="_blank">www.digium.com</a> & <a href="http://www.asterisk.org" rel="noreferrer noreferrer" target="_blank">www.asterisk.org</a><br>
<br>
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------------------------------<br>
<br>
Message: 3<br>
Date: Mon, 01 Jul 2019 11:57:45 -0300<br>
From: "Joshua C. Colp" <<a href="mailto:jcolp@digium.com" target="_blank" rel="noreferrer">jcolp@digium.com</a>><br>
To: <a href="mailto:asterisk-users@lists.digium.com" target="_blank" rel="noreferrer">asterisk-users@lists.digium.com</a><br>
Subject: Re: [asterisk-users] Second Asterisk server SIP JOIN a call<br>
to conduct a post-call survey<br>
Message-ID: <<a href="mailto:27f60411-06d6-4f75-a356-ca151e1f2505@www.fastmail.com" target="_blank" rel="noreferrer">27f60411-06d6-4f75-a356-ca151e1f2505@www.fastmail.com</a>><br>
Content-Type: text/plain<br>
<br>
On Mon, Jul 1, 2019, at 11:54 AM, Jason N wrote:<br>
> Unfortunately I am not allowed any changes to H's PBX / dialplan. <br>
> The restriction I have is that upon H's total disconnection from C, <br>
> that S continues the call with C. That's why I thought that if I could <br>
> get S to SIP JOIN the call from C, that once H disconnects S can <br>
> continue. I can extract the SIP call info on H and pass that to S (so <br>
> it can join the call). <br>
> <br>
> I'm just not sure if this concept is possible/practical.<br>
<br>
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.<br>
<br>
-- <br>
Joshua C. Colp<br>
Digium - A Sangoma Company | Senior Software Developer<br>
445 Jan Davis Drive NW - Huntsville, AL 35806 - US<br>
Check us out at: <a href="http://www.digium.com" rel="noreferrer noreferrer" target="_blank">www.digium.com</a> & <a href="http://www.asterisk.org" rel="noreferrer noreferrer" target="_blank">www.asterisk.org</a><br>
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New to Asterisk? Start here:<br>
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End of asterisk-users Digest, Vol 179, Issue 1<br>
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</blockquote></div>