On 18/09/15 08:54, Андрей Ярин wrote:

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/ 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.