18 Sep
2015
18 Sep
'15
4:54 p.m.
sr-users-request(a)lists.sip-router.org пишет:
Hello On 04/09/15 07:57, ?????? ???? wrote:
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"
8(351)7786878 доб 113
8(351)2555724 доб 113
8(351)2555787 доб 113
>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 
18 Sep
18 Sep
5:26 p.m.
New subject: Question about in call both side redirection
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");
>
22 Sep
22 Sep
4:04 p.m.
New subject: Question about in call both side redirection
On 18/09/15 08:54, Андрей Ярин wrote:
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.
/ 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
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days old
2 comments
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participants (2)
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Daniel-Constantin Mierla -
Андрей Ярин