[OpenSIPS-Users] Dispatcher stickyness for external inbound calls
David Villasmil
david.villasmil.work at gmail.com
Fri Dec 6 05:34:55 EST 2019
Hello,
I totally misunderstood your question, then.
Still dialplan should be useful. Instead of matching the domain, match the
$ru and send it where it is supposed to be registered based on the
dispatcher id.
If the user can only register on one fs, then you’re good for registers and
conferences.
If the user is allowed to use multiple fs, you need to devise some way of
figuring out where the conference is taking place (maybe send the invite to
all the fs The user is allowed to, if you get a not found for the dialed
number (conference), send it to next one, eventually it will find the
conference, if it exists.
How I do it is I create the conference on fs with the first user, and
instruct the second user (via wss and jsip as the user is logged into our
web page) to call into the conference number and adding a header specifying
the server id, so that the proxy doesn’t need to do any lookups. The proxy
looks into the header, translates the header value into a IP address, and
forward it to the fs server.
Having the user register on fs when you have OpenSIPS/kamailio only
complicates things, not to mention OpenSIPS can take A LOT more registers
than fs ( on another setup I had around 8k users and eventually moved to
Kamailio/OpenSIPS as a registrar, this increased considerably and OS/K was
just fine, barely CPU usage). Your best bet is to create your necessary
logic for registers on OS/K and use fs only for conferences.
Hope that helps!
David
On Fri, 6 Dec 2019 at 07:17, Matthias Kneer via Users <
users at lists.opensips.org> wrote:
> Thanks for your reply and the example, David.
>
> If I did understand your example correct, it basically solves the issue
> only for internal calls because of the match for ".*domain*"
>
> My main issue is the scenario where I do get a call from the PSTN. Those
> calls do not contain the corresponding FreeSWITCH domain of an
> extension, they do only contain the floating IP of my openSIPS systems
> and the external number which has being called. This means, that I can't
> match for the domain. This is how an INVITE from the PSTN looks like for
> me:
>
> INVITE sip:+49691234567 at 1.2.3.4:5060 SIP/2.0
> Via: SIP/2.0/UDP 1.2.3.5:5060;branch=z9hG4bK06Ba5515bda5556f434
> From: <sip:+491751231231 at 1.2.3.5>;tag=gK06634aaa
> To: <sip:+49691234567 at 1.2.3.4>
> Call-ID: 1931912619_117228776 at 1.2.3.5
> CSeq: 590408 INVITE
> [...]
>
> and this is how an internal INVITE looks for me:
>
> INVITE sip:567 at customer1.sip.example.org;transport=TCP SIP/2.0
> Via: SIP/2.0/TCP
> 192.168.1.168:32396;branch=z9hG4bK-524287-1---62375638a60d1cee;rport
> Max-Forwards: 70
> Contact: <sip:789 at 1.2.2.3:34511;transport=TCP>
> To: <sip:567 at customer1.sip.example.org>
> From: <sip:789 at customer1.sip.example.org;transport=TCP>;tag=1b546d48
> Call-ID: Yr7uhzwS-LFx5hKjB8In7g..
> CSeq: 1 INVITE
> [...]
>
> What I would need is the related domain within FreeSWITCH of the number
> +49691234567 so that it will then be sent to the same FS where this
> device has been registered with its internal extension / where the other
> conference calls have been sent to. But since the device registers with
> e.g. 567 at customer1.sip.example.org which does not directly relate to
> +49691234567 at 1.2.3.4, openSIPS is not aware of the FreeSWITCH where this
> device was registered / the other conference calls where sent to.
>
> A simple logic I was thinking about was something like this:
>
> $var(customdom) = $fd
>
> if ($rU=~"\+49691234.*") {
> $var(customdom) = "customer1.sip.example.org"
> }
>
> later backed by a database so that we not have to hardcode this into the
> routing logic. Would this also be something for the dialplan?
>
> Thanks,
> Matthias
>
>
> Am 05.12.2019 18:02, schrieb David Villasmil:
> > Hello
> >
> > Dialplan is perfect for this.
> > I did this on kamailio, but it's probably doable porting it.
> >
> > It would be something like:
> >
> > modparam("dialplan", "db_url", DBURL)
> >
> > ...
> >
> > route[DISPATCH] {
> > ...
> > # Get the attrs field for the matching domain, will store the result
> > in $var(dispatcher_id)
> > dp_translate( "1", "$ru", "$var(dispatcher_id)" )
> >
> > # Select the group of fs nodes that we want this call to be sent to.
> > if(!ds_select_dst("$var(dispatcher_id)", "4"))
> > {
> > xlog("[DISPATCH]: ds_select_dst FAILED!'\n");
> > send_reply("404", "No destination");
> > exit;
> > }
> >
> > xlog("L_DBG", "[DISPATCH]: going to <$ru> via <$du>\n");
> > t_on_failure("RTF_DISPATCH");
> > route(RELAY);
> > }
> >
> > Your dialplan table would look like this:
> >
> >
> +----+------+-----+----------+---------------------------+-----------+-----------+----------+-------+
> > | id | dpid | pr | match_op | match_exp | match_len |
> > subst_exp | repl_exp | attrs |
> >
> +----+------+-----+----------+---------------------------+-----------+-----------+----------+-------+
> > | 37 | 1 | 100 | 1 | .*domain1* | 0 |
> > | | 10 |
> > | 55 | 1 | 100 | 1 | .*domain2* | 0 |
> > | | 30 |
> > | 56 | 1 | 100 | 1 | .*domain3* | 0 |
> > | | 30 |
> >
> > and your dispatcher would look like you want to distribute your
> > traffic:
> >
> >
> +------+-------+-------------------------+-------+----------+----------------------------------+------------------+
> > | id | setid | destination | flags | priority | attrs
> > | description |
> >
> +------+-------+-------------------------+-------+----------+----------------------------------+------------------+
> > | 3097 | 10 | sip:1.2.3.1:5080 [1] | 0 | 10 |
> > | server-01 |
> > | 3098 | 10 | sip:1.2.3.2:5080 [2] | 0 | 10 |
> > | server-02 |
> > | 3110 | 20 | sip:1.2.3.1:5080 [1] | 0 | 10 |
> > | server-01 |
> > | 3109 | 20 | sip:1.2.3.2:5080 [2] | 0 | 10 |
> > | server-02 |
> > | 3108 | 30 | sip:1.2.3.3:5080 [3] | 0 | 10 |
> > | server-03 |
> >
> > I hope that helps!
> >
> > David
> >
> > On Thu, 5 Dec 2019 at 15:01, Matthias Kneer via Users
> > <users at lists.opensips.org> wrote:
> >
> >> Hi list,
> >>
> >> I'm currently building a HA setup with 2 openSIPS 2.4 instances
> >> which do share a floating IP through keepalived. They are working as
> >> 1:1 mid_registrar, gateway to the PSTN and dispatcher for multiple
> >> FreeSWITCH instances. Those FreeSWITCH instances hold all accounts,
> >> domains and media services, they also do the RTP part.
> >>
> >> Since we are hosting multiple customers on the same FreeSWITCH
> >> instances, sperated through domains, we'd like to ensure that all
> >> registrations of a domain stay on the same FreeSWITCH as long this
> >> instance is reachable. We want this to ensure that conferences, call
> >> pickups and so on work reliable. It's not very important for us to
> >> distribute the load evenly since we have no high call / conference
> >> volume, the goal is more to have the setup as available as possible
> >> with some sort of loadbalancing and without having to take manual
> >> action if one of the FreeSWITCHes dies.
> >>
> >> To achieve this, our idea was to set modparam("dispatcher",
> >> "hash_pvar", "$fd") to use the from domain as source for hashing and
> >> call ds_select_dst with the alg "7":
> >>
> >> if (!ds_select_dst("2", "7"))
> >> {
> >> xlog("L_ERR", "$ci|end|no servers avaliable");
> >> sl_send_reply("480", "Temporarily Unavailable");
> >> exit;
> >> }
> >>
> >> This approach works fine as long as we use it for internal calls
> >> since those have the from / to domain part properly set. If a call
> >> arrives from the PSTN openSIPS has no knowledge about the relation
> >> of the external numbers and internal numbers / domains. Is there a
> >> smart way to properly route those external calls to the correct
> >> FreeSWITCH wihtout having openSIPS to access the FreeSWITCH database
> >> / duplicating the data into the openSIPS database or to run
> >> FreeSWITCH in active/standby mode?
> >>
> >> Best regards,
> >> Matthias
> >> _______________________________________________
> >> Users mailing list
> >> Users at lists.opensips.org
> >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >
> >
> > Links:
> > ------
> > [1] http://1.2.3.1:5080
> > [2] http://1.2.3.2:5080
> > [3] http://1.2.3.3:5080
> > _______________________________________________
> > Users mailing list
> > Users at lists.opensips.org
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
--
Regards,
David Villasmil
email: david.villasmil.work at gmail.com
phone: +34669448337
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