On Mon, Nov 14, 2011 at 10:08 AM, Vik Killa <vipkilla@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@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@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@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@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@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@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@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@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@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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>
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