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

[Андрей Ярин]

> On 18/09/15 08:54, Андрей Ярин wrote:
> >/  sr-users-request at lists.sip-router.org  <http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>  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 at 10.10.9.209  <http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>");
> />>>/  >dlg_refer("caller","sip:100 at 10.10.9.209  <http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>");
> />>>/  >
> />>/  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 Mierlahttp://twitter.com/#!/miconda  <http://twitter.com/#%21/miconda>  -
> />>/  http://www.linkedin.com/in/miconda  Book: SIP Routing With Kamailio -
> />>/  http://www.asipto.com  <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
I try to impliment HA in link Kamailio-Asterisk. Without asterisk if one 
server down, dialog continues on other server (keepalived + DB mirror), 
direct RTP.
But we use voice services (transcoding, voicemail and others), so i need 
add asterisk to dialog. Main problem RTP - if asterisk down, rtp goes 
nowhere. The idea is - kamailio will tell both sides whitch server to 
use, and redirect to other server when main fails and users will not 
notice server problems. Skypelike behavior.
Because i cant tell asterisk how to process RTP without SIP (i think 
adding H.248/MEGACO support to kamailio will be useful), so i need to 
redirect both sides to dynamic number, which will be conference room for 
2 users at asterisk. Or voicemail. Or something else.