[Freeswitch-users] Setup FreeSwitch behind Kamailio Dispatcher.

Leon de Rooij leon at scarlet-internet.nl
Tue Nov 15 17:05:12 MSK 2011 

Hello Sammy,

Well, Kamailio and OpenSIPS are essentially the same, eh. They are both forks from the same codebase.

Kind regards,

Leon


On Nov 15, 2011, at 12:33 PM, Sammy Govind wrote:

> Looking at the original topic and the opensips.cfg really confused me that we started from kamailio and ended up in openSIPS ! 
> 
> On Tue, Nov 15, 2011 at 3:50 PM, Leon de Rooij <leon at scarlet-internet.nl> wrote:
> Hello Vik and Matt,
> 
> Of course, here's my current opensips.cfg, but note that it contains bugs and is far from ready, just call setup/teardown and registrations work.
> 
> Also, I have a dispatcher.list containing one set ("1") pointing to two sip profiles (at different udp ports) on one fs instance. Both sip profiles have the outbound-proxy param set to the sip uri of the proxy.
> 
> kind regards,
> 
> Leon
> 
> 
> 
> 
> 
> On Nov 14, 2011, at 5:18 PM, Matt Stockton wrote:
> 
>> I would love to see the config too. I'm actively trying to use OpenSips as an inbound / outbound proxy SBC and am having mixed success as well
>> 
>> On Mon, Nov 14, 2011 at 10:08 AM, Vik Killa <vipkilla at gmail.com> wrote:
>> Any chance you can post your opensips config? I've been struggling
>> with mixed success trying same setup.
>> 
>> On Mon, Nov 14, 2011 at 10:18 AM, Leon de Rooij
>> <leon at scarlet-internet.nl> wrote:
>> > Hi,
>> > I'm actually trying to configure the same thing right now - with opensips
>> > though, but (afaik) it uses the same dispatcher module.
>> > First I used a simple route script in opensips with using dispatcher, but
>> > after the first message (from ua through proxy to fs), the proxy would get
>> > out of the signaling path, while I want it to stay in.
>> > To fix that, I added record routing in the proxy configuration so it stays
>> > in the path.
>> > Registrations are also balanced towards the fs servers - but some clients
>> > are nat'ed so the contact header was wrong - fix_nated_contact() in route
>> > and in  onreply_route fixed that.
>> > To get calls originating from fs to go through the proxy before going to the
>> > client, I tried:
>> > originate
>> > sofia/some_profile/sip:user at mydomain.com;fs_path=sip:ip_of_sip_proxy:5060
>> > &park
>> > which worked. Then I found out about the sip_route_uri variable that can be
>> > set in the bridge string:
>> > originate
>> > {sip_route_uri=sip:ip_of_sip_proxy:5060}sofia/some_profile/sip:some_user at mydomain.com
>> > &park
>> > which seemed to do the same thing (does anyone know if there's any
>> > difference between the two variants ?)
>> > Then I tried putting the sip_route_uri in the dial-string param of the
>> > domain (or user), for example:
>> > <param name="dial-string"
>> > value="{sip_route_uri=sip:ip_of_sip_proxy:5060,sip_invite_domain=${dialed_domain},presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
>> > so now it's possible to just call the user as:
>> > originate user/some_user at mydomain.com &park
>> > nice !
>> > And then I found out that you can just set an outbound-proxy param in the
>> > sip profile that's used for originating:
>> > <param name="outbound-proxy" value="sip:ip_of_sip_proxy:5060"/>
>> > so now you can leave out the sip_route_uri or fs_path variables, it just
>> > works.
>> > My proxy config still needs a lot of work, but I got the basic functionality
>> > working, still need to find out about using acls on fs side based on a sip
>> > header added by the proxy - I'm already adding X-AUTH-IP headers on messages
>> > from clients towards fs which should do the trick, but I didn't test it yet.
>> > Anyway, would love to read some configs of ppl who successfuly setup
>> > opensips/kamailio/openser/... proxies in front of fs, what their experience
>> > is with using record-routing or not, etc.
>> > regards,
>> > Leon
>> >
>> >
>> >
>> > On Nov 14, 2011, at 6:40 AM, Henrik Aagaard Sørensen wrote:
>> >
>> > The handling of several FS are not a issue, actually easy enough.
>> > I would like to have as little load on Kamailio as possible, as it just
>> > should load balance.
>> > Also, having to handle users on both Kamailio and FS makes unecessary work
>> > loads.
>> >
>> > On 14/11/2011, at 00.32, Sammy Govind <govoiper at gmail.com> wrote:
>> >
>> > Hi again,
>> > Why don't you just let Kamailio handle registrations. Anyway I was thinking
>> > about that LBing the registration would result in such a scenario that
>> > calling one extension to another would not make a successful call because
>> > the other endpoint maybe registered on some other FS.
>> > This may further lead you to making a dial-plan which would work somewhat
>> > like DUNDI but it'd just have to search all the FS servers before joining a
>> > call.(Obv there are other intelligent approaches to minimize the headache)
>> > Try following the link I sent you and implement that in front of your FS, I
>> > think that Kamailio configuration is so well written that anyone can start
>> > understanding kamailio and implement such setups with little effort.
>> > By explaining your topology I meant how do you plan to use Kamailio in front
>> > of FS? i.e Kamailio on Public IP and all FS on private IPs and etc  as in a
>> > topo-hiding or SBC like setup!
>> > --
>> > Regards,
>> > Sammy
>> >
>> > 2011/11/14 Henrik Aagaard Sørensen <henrikaagaardsorensen at gmail.com>
>> >>
>> >> Hi Sammy.
>> >> I've actually removed registration and presence from Kamailio, so all it
>> >> does is dispatch everything to FreeSwitch.
>> >> Currently I only have 1 FreeSwitch, for testing this basic setup.
>> >> Next move would be 2 FreeSwitch with 1 common database etc. But that's
>> >> later.
>> >> My FreeSwitch is the basic setup, without anything else. Just as the
>> >> installation manuel is written. So I have my extension 1000 - 1019 etc. And
>> >> calls between them works when connected directly to FreeSwitch.
>> >> But when going through Kamailio Dispatcher it fails between the
>> >> extensions.
>> >> So I guess there should be some more setup in FreeSwitch when using a load
>> >> balancer (dispatcher) in front of it.
>> >> My setup is just Kamailio -> FreeSwitch and FreeSwitch should handle
>> >> everything.
>> >>
>> >> On Mon, Nov 14, 2011 at 5:56 AM, Sammy Govind <govoiper at gmail.com> wrote:
>> >>>
>> >>> Hi,
>> >>> If everything is setup as you expected it then don't read this
>> >>> > http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb,
>> >>> else follow the article from start till end and implement it. Don't worry
>> >>> about the name asterisk, just replace that all with FreeSWICTH. It will work
>> >>> like that too.
>> >>> Once that is implemented then apply dispatcher module on the route which
>> >>> says [REGFWD] or [TOASTERISK].
>> >>> Also explain your topology abit more  and features required as well that
>> >>> may help in telling which extra module you will require in order to make
>> >>> things work.
>> >>> In your current implementation are you sure SIP phones are registering on
>> >>> FS and not on Kamailio?and that both end points making call to each other
>> >>> are on same FS?
>> >>> --
>> >>> Regards,
>> >>> Sammy
>> >>>
>> >>> 2011/11/14 Henrik Aagaard Sørensen <henrikaagaardsorensen at gmail.com>
>> >>>>
>> >>>> Hi everyone. In regards to my earlier question regarding with FreeSwitch
>> >>>> behind Kamailio Dispatcher, I've attached a call from extension 1001 to
>> >>>> 1002, which fails. It just hangs for some time and then says that 1002
>> >>>> cannot be found, and then the voicemail for it comes up.
>> >>>> 2011/11/13 Henrik Aagaard Sørensen <henrikaagaardsorensen at gmail.com>
>> >>>>>
>> >>>>> I'm trying to get the setup Kamailio Dispatcher -> FreeSwitch to work.
>> >>>>> I've setup Kamailio via
>> >>>>> this: http://www.kamailio.org/docs/modules/stable/modules_k/dispatcher.html
>> >>>>> I've installed FreeSwitch from scratch on Ubuntu
>> >>>>> via: http://wiki.freeswitch.org/wiki/Ubuntu_Quick_Start
>> >>>>> Now, when registering extensions via Kamailio Dispatcher I'm able to
>> >>>>> call to FreeSwitch and listen to hold music. But that's it. I'm not able to
>> >>>>> call between extensions etc.
>> >>>>> Can anyone help me setting up FreeSwitch to accept registration, calls
>> >>>>> etc. from Kamailio and everything else that is needed to use FreeSwitch
>> >>>>> behind a load balancer?
>> >>>>> I'm very new to FreeSwitch, but I'm trying to use the terminal (without
>> >>>>> any GUI etc.) as I want the installation to be as clean as possible. So I
>> >>>>> would prefer very precise help, as I'm still getting hold of FreeSwitch.
>> >>>>
>> >>>>
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