[SR-Users] Question about in call both side redirection

[Daniel-Constantin Mierla]

Hello

On 04/09/15 07:57, Андрей Ярин wrote:
> Hello (sorry for my bad english) - i try to create voice record
> service by request. User A call to user B. In call by pressing
> combination like *55 Kamailio must redirect both sides to asterisk,
> whitch create dynamic conference room with recording. As i understand
> i need to use dlg_refer() from dialog module, but in log file i get:
> Konsole output
> Sep  4 10:45:14 voipc-node2 /usr/local/sbin/kamailio[18941]: ERROR:
> dialog [dlg_req_within.c:85]: build_dlg_t(): no contact available
> Sep  4 10:45:14 voipc-node2 /usr/local/sbin/kamailio[18941]: ERROR:
> dialog [dlg_transfer.c:188]: dlg_refer_callee(): failed to create dlg_t
>
>
> In script i try to refer with:
> dlg_refer("callee", "sip:100 at 10.10.9.209");
> dlg_refer("caller", "sip:100 at 10.10.9.209");
>
in what context do you use the above actions? In other words, do you
execute them when you process a specific request? If yes, which one?

Another question, how do you capture when *55 is pressed? Is dtmf sent
via sip info request?

Cheers,
Daniel

-- 
Daniel-Constantin Mierla
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