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 przeqpiciel@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 abalashov@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@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@evaristesys.com:
It can record, as can a number of other media relays.
On March 14, 2017 2:43:15 AM EDT, przeqpiciel
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@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@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
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,
>>> >>> 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@evaristesys.com
>>>> mailto:abalashov@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/,
>>>> >>>> _______________________________________________ >>>> 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
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-- Alex
-- Principal, Evariste Systems LLC (www.evaristesys.com)
Sent from my Google Nexus.
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
list
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-- Alex
-- Principal, Evariste Systems LLC (www.evaristesys.com)
Sent from my Google Nexus.