18 Sep
2015
18 Sep
'15
5:26 p.m.
On 18/09/15 08:54, Андрей Ярин wrote:
sr-users-request(a)lists.sip-router.org пишет:
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
--
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
Hello On 04/09/15 07:57, ?????? ???? wrote:
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
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");
>