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

[Андрей Ярин]

sr-users-request at 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 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 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

-- 
Ярин Андрей
Инженер ООО "И.С.-Телеком"
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