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.
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
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
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
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
