[Freeswitch-users] Setup FreeSwitch behind Kamailio Dispatcher.

Leon de Rooij leon at scarlet-internet.nl
Mon Nov 14 18:18:29 MSK 2011 

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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