An ambitious endeavour. Are you sure it's an economically sensible one, given 
that there are a variety of solutions already out there? 

On March 14, 2017 2:55:31 AM EDT, przeqpiciel <przeqpic...@gmail.com> wrote:
>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 <abalas...@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 <przeqpic...@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 <abalas...@evaristesys.com>:
>> >
>> >> It can record, as can a number of other media relays.
>> >>
>> >> On March 14, 2017 2:43:15 AM EDT, przeqpiciel
><przeqpic...@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 <anfec...@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 <k...@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
><k...@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"
><abalas...@evaristesys.com
>> >> >>>>>> <mailto:abalas...@evaristesys.com>> napisaƂ(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@lists.sip-router.org
>> >> >>>>>>
>> >> >>>>> <mailto:sr-users@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@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@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@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@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@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@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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