[Freeswitch-users] Setup FreeSwitch behind Kamailio Dispatcher.

Sammy Govind govoiper at gmail.com
Tue Nov 15 18:06:03 MSK 2011 

:) :) :)
LOL-- right- Sorry, I'm from Narnia didn't know.

Original Poster had issue with Kamailio and FS setup, even if both the
projects share common parents but I don't believe that sharing opensips.cfg
will help the issue here. Anyways I appreciate the spirit to help others.

--
BR,
Sammy


On Tue, Nov 15, 2011 at 7:05 PM, Leon de Rooij <leon at scarlet-internet.nl>wrote:

> 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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