[SR-Users] Several Asterisk on this same IP

Tue Mar 14 07:55:31 CET 2017

I would like to create PBX platform, at now I faced to make drag&drop ivr
creator. After that I would create option for record calls for client and
this is why I look for solution :)

2017-03-14 7:47 GMT+01:00 Alex Balashov <abalashov at evaristesys.com>:

> Yes, though of course you would have to correlate the calls (most likely
> by Call-ID) and integrate all this.
>
>
> On March 14, 2017 2:46:27 AM EDT, przeqpiciel <przeqpiciel at gmail.com>
> wrote:
> >So, I can use Kamailio as SBC/Load balancer/registrar, Asterisk as IVR
> >and
> >application server, and rtpproxy as media relay and recorder ?
> >
> >2017-03-14 7:44 GMT+01:00 Alex Balashov <abalashov at evaristesys.com>:
> >
> >> It can record, as can a number of other media relays.
> >>
> >> On March 14, 2017 2:43:15 AM EDT, przeqpiciel <przeqpiciel at gmail.com>
> >> wrote:
> >> >>> WHy not installing rtpproxy and proxying all
> >> >Because I would like to record some calls and I dont know RTPProxy's
> >> >features, maybe it could record ?
> >> >
> >> >2017-03-14 5:14 GMT+01:00 anfecora <anfecora at gmail.com>:
> >> >
> >> >> WHy not installing rtpproxy and proxying all rtp to the inside
> >uase
> >> >> kamailio to load balance them, it will be transparent on the
> >inside
> >> >perhaps
> >> >> a cleaner solution?
> >> >>
> >> >> On Mon, Mar 13, 2017 at 3:21 PM, Kjeld Flarup <kfc at viptel.dk>
> >wrote:
> >> >>
> >> >>> As I recall it is sequential, but not from the start everytime,
> >it
> >> >is
> >> >>> incrementing all the time.
> >> >>>
> >> >>> If You are running three servers, then with a 100% identical
> >load,
> >> >one
> >> >>> would expect an average of 2 failing attempts per call.
> >> >>>
> >> >>> The reality I see is however often very different RTP ports, most
> >> >likely
> >> >>> because load isn't 100% identical.
> >> >>>
> >> >>>
> >> >>> Med venlig hilsen / Best regards
> >> >>> Kjeld Flarup (Christensen) M.Sc E.E, Teknisk chef
> >> >>> Viptel ApS, Hammershusvej 16C, DK-7400 Herning
> >> >>> Telefon: +45 46949949, Telefax: +45 46949950, http://viptel.dk
> >> >>>
> >> >>> On 03/13/2017 11:05 PM, Alex Balashov wrote:
> >> >>>
> >> >>>> Well, indeed, but a sequential scan of many consecutive ports
> >like
> >> >this
> >> >>>> from the bottom of the same range can be quite a latent
> >operation.
> >> >So at
> >> >>>> the very least the allocation strategy would benefit from being
> >> >random.
> >> >>>> Does Asterisk take that approach?
> >> >>>>
> >> >>>> On March 13, 2017 6:04:06 PM EDT, Kjeld Flarup <kfc at viptel.dk>
> >> >wrote:
> >> >>>>
> >> >>>>> No there is no such thing as magic.
> >> >>>>>
> >> >>>>> The most obvious way to implement the RTP port handling, is to
> >> >first
> >> >>>>> open the next UDP port in the OS, and then report that back in
> >the
> >> >>>>> Invite/200Ok. If the port cannot be opened, then simply try the
> >> >next in
> >> >>>>>
> >> >>>>> line.
> >> >>>>>
> >> >>>>>
> >> >>>>> Med venlig hilsen / Best regards
> >> >>>>> Kjeld Flarup (Christensen) M.Sc E.E, Teknisk chef
> >> >>>>> Viptel ApS, Hammershusvej 16C, DK-7400 Herning
> >> >>>>> Telefon: +45 46949949, Telefax: +45 46949950, http://viptel.dk
> >> >>>>>
> >> >>>>> On 03/13/2017 01:52 PM, przeqpiciel wrote:
> >> >>>>>
> >> >>>>>> Maybe there is an magic device? I know that if we have an
> >> >asterisk,
> >> >>>>>> that become to us with default configuration of rtp ports sets
> >to
> >> >>>>>> 10000_20000. And each call choose the one port fron that
> >range.
> >> >So if
> >> >>>>>> we have several asterisks with default configuratiin of rtp,
> >> >there is
> >> >>>>>> possibilities to have 2 concurent calls each through another
> >> >asterisk
> >> >>>>>> instance with this same rtp port. Am i right?
> >> >>>>>>
> >> >>>>>> So mqybe this magic device could see source IP address and
> >route
> >> >rtp
> >> >>>>>> to correct adterisk?
> >> >>>>>>
> >> >>>>>> 13.03.2017 7:15 AM "Alex Balashov" <abalashov at evaristesys.com
> >> >>>>>> <mailto:abalashov at evaristesys.com>> napisał(a):
> >> >>>>>>
> >> >>>>>>      On Mon, Mar 13, 2017 at 07:08:09AM +0100, Kjeld Flarup
> >> >wrote:
> >> >>>>>>
> >> >>>>>>      > We run multiple Asterisk instances since 1.4 and never
> >> >>>>>>      configured RTP ports.
> >> >>>>>>      >
> >> >>>>>>      > More challenging issues are the Asterisk DB, and the
> >> >Asteisk
> >> >>>>>>
> >> >>>>> home.
> >> >>>>>
> >> >>>>>>      You may not have enough calls for RTP port collisions to
> >> >become
> >> >>>>>>
> >> >>>>> an
> >> >>>>>
> >> >>>>>>      issue. Otherwise, I'm not sure how you're avoiding it,
> >since
> >> >>>>>>
> >> >>>>> Asterisk
> >> >>>>>
> >> >>>>>>      isn't aware of which ports from within the range are in
> >use.
> >> >>>>>>
> >> >>>>>>      --
> >> >>>>>>      Alex Balashov | Principal | Evariste Systems LLC
> >> >>>>>>
> >> >>>>>>      Tel: +1-706-510-6800 <tel:%2B1-706-510-6800> /
> >> >+1-800-250-5920
> >> >>>>>>      <tel:%2B1-800-250-5920> (toll-free)
> >> >>>>>>      Web: http://www.evaristesys.com/,
> >http://www.csrpswitch.com/
> >> >>>>>>
> >> >>>>>>      _______________________________________________
> >> >>>>>>      SIP Express Router (SER) and Kamailio (OpenSER) -
> >sr-users
> >> >>>>>>
> >> >>>>> mailing
> >> >>>>>
> >> >>>>>>      list
> >> >>>>>>      sr-users at lists.sip-router.org
> >> >>>>>>
> >> >>>>> <mailto:sr-users at lists.sip-router.org>
> >> >>>>>
> >> >>>>>>
> >> >http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
> >> >>>>>>
> >> ><http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>
> >> >>>>>>
> >> >>>>>>
> >> >>>>>>
> >> >>>>>> _______________________________________________
> >> >>>>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users
> >> >mailing
> >> >>>>>>
> >> >>>>> list
> >> >>>>>
> >> >>>>>> sr-users at lists.sip-router.org
> >> >>>>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
> >> >>>>>>
> >> >>>>>
> >> >>>> -- Alex
> >> >>>>
> >> >>>> --
> >> >>>> Principal, Evariste Systems LLC (www.evaristesys.com)
> >> >>>>
> >> >>>> Sent from my Google Nexus.
> >> >>>>
> >> >>>> _______________________________________________
> >> >>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users
> >mailing
> >> >list
> >> >>>> sr-users at lists.sip-router.org
> >> >>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
> >> >>>>
> >> >>>
> >> >>>
> >> >>> _______________________________________________
> >> >>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users
> >mailing
> >> >list
> >> >>> sr-users at lists.sip-router.org
> >> >>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
> >> >>>
> >> >>
> >> >>
> >> >> _______________________________________________
> >> >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
> >> >list
> >> >> sr-users at lists.sip-router.org
> >> >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
> >> >>
> >> >>
> >>
> >>
> >> -- Alex
> >>
> >> --
> >> Principal, Evariste Systems LLC (www.evaristesys.com)
> >>
> >> Sent from my Google Nexus.
> >>
> >> _______________________________________________
> >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
> >list
> >> sr-users at lists.sip-router.org
> >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
> >>
>
>
> -- Alex
>
> --
> Principal, Evariste Systems LLC (www.evaristesys.com)
>
> Sent from my Google Nexus.
>
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