On 18/09/15 08:54, Андрей Ярин wrote:
sr-users-request@lists.sip-router.org пишет:
Hello On 04/09/15 07:57, ?????? ???? 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@10.10.9.209"); dlg_refer("caller","sip:100@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 http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com Kamailio Advanced Training, Sep 28-30, 2015, in Berlin - http://asipto.com/u/kat
For now i try to use event_route[dialog:start] - i testing - can kamailio redirect both sides to external service, and will it work with event_route[dispatcher:dst-down]. If it will work, i will add SIP INFO processing for service codes
I don't get the context of involving event_route[dispatcher:dst-down], maybe you can present with more details how you plan to do the whole thing.
Cheers, Daniel