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>> 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@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 >
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