[OpenSIPS-Users] Dispatcher stickyness for external inbound calls
Matthias Kneer
matze at roterruler.de
Fri Dec 6 02:17:09 EST 2019
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
>> _______________________________________________
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>> 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
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