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<br><br><div class="gmail_quote">On Mon, Nov 14, 2011 at 10:08 AM, Vik Killa <span dir="ltr"><<a href="mailto:vipkilla@gmail.com">vipkilla@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">Any chance you can post your opensips config? I've been struggling<br>
with mixed success trying same setup.<br>
<div class="HOEnZb"><div class="h5"><br>
On Mon, Nov 14, 2011 at 10:18 AM, Leon de Rooij<br>
<<a href="mailto:leon@scarlet-internet.nl">leon@scarlet-internet.nl</a>> wrote:<br>
> Hi,<br>
> I'm actually trying to configure the same thing right now - with opensips<br>
> though, but (afaik) it uses the same dispatcher module.<br>
> First I used a simple route script in opensips with using dispatcher, but<br>
> after the first message (from ua through proxy to fs), the proxy would get<br>
> out of the signaling path, while I want it to stay in.<br>
> To fix that, I added record routing in the proxy configuration so it stays<br>
> in the path.<br>
> Registrations are also balanced towards the fs servers - but some clients<br>
> are nat'ed so the contact header was wrong - fix_nated_contact() in route<br>
> and in onreply_route fixed that.<br>
> To get calls originating from fs to go through the proxy before going to the<br>
> client, I tried:<br>
> originate<br>
> sofia/some_profile/<a href="mailto:sip%3Auser@mydomain.com">sip:user@mydomain.com</a>;fs_path=sip:ip_of_sip_proxy:5060<br>
> &park<br>
> which worked. Then I found out about the sip_route_uri variable that can be<br>
> set in the bridge string:<br>
> originate<br>
> {sip_route_uri=sip:ip_of_sip_proxy:5060}sofia/some_profile/<a href="mailto:sip%3Asome_user@mydomain.com">sip:some_user@mydomain.com</a><br>
> &park<br>
> which seemed to do the same thing (does anyone know if there's any<br>
> difference between the two variants ?)<br>
> Then I tried putting the sip_route_uri in the dial-string param of the<br>
> domain (or user), for example:<br>
> <param name="dial-string"<br>
> 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})}"/><br>
> so now it's possible to just call the user as:<br>
> originate user/<a href="mailto:some_user@mydomain.com">some_user@mydomain.com</a> &park<br>
> nice !<br>
> And then I found out that you can just set an outbound-proxy param in the<br>
> sip profile that's used for originating:<br>
> <param name="outbound-proxy" value="sip:ip_of_sip_proxy:5060"/><br>
> so now you can leave out the sip_route_uri or fs_path variables, it just<br>
> works.<br>
> My proxy config still needs a lot of work, but I got the basic functionality<br>
> working, still need to find out about using acls on fs side based on a sip<br>
> header added by the proxy - I'm already adding X-AUTH-IP headers on messages<br>
> from clients towards fs which should do the trick, but I didn't test it yet.<br>
> Anyway, would love to read some configs of ppl who successfuly setup<br>
> opensips/kamailio/openser/... proxies in front of fs, what their experience<br>
> is with using record-routing or not, etc.<br>
> regards,<br>
> Leon<br>
><br>
><br>
><br>
> On Nov 14, 2011, at 6:40 AM, Henrik Aagaard Sørensen wrote:<br>
><br>
> The handling of several FS are not a issue, actually easy enough.<br>
> I would like to have as little load on Kamailio as possible, as it just<br>
> should load balance.<br>
> Also, having to handle users on both Kamailio and FS makes unecessary work<br>
> loads.<br>
><br>
> On 14/11/2011, at 00.32, Sammy Govind <<a href="mailto:govoiper@gmail.com">govoiper@gmail.com</a>> wrote:<br>
><br>
> Hi again,<br>
> Why don't you just let Kamailio handle registrations. Anyway I was thinking<br>
> about that LBing the registration would result in such a scenario that<br>
> calling one extension to another would not make a successful call because<br>
> the other endpoint maybe registered on some other FS.<br>
> This may further lead you to making a dial-plan which would work somewhat<br>
> like DUNDI but it'd just have to search all the FS servers before joining a<br>
> call.(Obv there are other intelligent approaches to minimize the headache)<br>
> Try following the link I sent you and implement that in front of your FS, I<br>
> think that Kamailio configuration is so well written that anyone can start<br>
> understanding kamailio and implement such setups with little effort.<br>
> By explaining your topology I meant how do you plan to use Kamailio in front<br>
> of FS? i.e Kamailio on Public IP and all FS on private IPs and etc as in a<br>
> topo-hiding or SBC like setup!<br>
> --<br>
> Regards,<br>
> Sammy<br>
><br>
> 2011/11/14 Henrik Aagaard Sørensen <<a href="mailto:henrikaagaardsorensen@gmail.com">henrikaagaardsorensen@gmail.com</a>><br>
>><br>
>> Hi Sammy.<br>
>> I've actually removed registration and presence from Kamailio, so all it<br>
>> does is dispatch everything to FreeSwitch.<br>
>> Currently I only have 1 FreeSwitch, for testing this basic setup.<br>
>> Next move would be 2 FreeSwitch with 1 common database etc. But that's<br>
>> later.<br>
>> My FreeSwitch is the basic setup, without anything else. Just as the<br>
>> installation manuel is written. So I have my extension 1000 - 1019 etc. And<br>
>> calls between them works when connected directly to FreeSwitch.<br>
>> But when going through Kamailio Dispatcher it fails between the<br>
>> extensions.<br>
>> So I guess there should be some more setup in FreeSwitch when using a load<br>
>> balancer (dispatcher) in front of it.<br>
>> My setup is just Kamailio -> FreeSwitch and FreeSwitch should handle<br>
>> everything.<br>
>><br>
>> On Mon, Nov 14, 2011 at 5:56 AM, Sammy Govind <<a href="mailto:govoiper@gmail.com">govoiper@gmail.com</a>> wrote:<br>
>>><br>
>>> Hi,<br>
>>> If everything is setup as you expected it then don't read this<br>
>>> > <a href="http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb" target="_blank">http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb</a>,<br>
>>> else follow the article from start till end and implement it. Don't worry<br>
>>> about the name asterisk, just replace that all with FreeSWICTH. It will work<br>
>>> like that too.<br>
>>> Once that is implemented then apply dispatcher module on the route which<br>
>>> says [REGFWD] or [TOASTERISK].<br>
>>> Also explain your topology abit more and features required as well that<br>
>>> may help in telling which extra module you will require in order to make<br>
>>> things work.<br>
>>> In your current implementation are you sure SIP phones are registering on<br>
>>> FS and not on Kamailio?and that both end points making call to each other<br>
>>> are on same FS?<br>
>>> --<br>
>>> Regards,<br>
>>> Sammy<br>
>>><br>
>>> 2011/11/14 Henrik Aagaard Sørensen <<a href="mailto:henrikaagaardsorensen@gmail.com">henrikaagaardsorensen@gmail.com</a>><br>
>>>><br>
>>>> Hi everyone. In regards to my earlier question regarding with FreeSwitch<br>
>>>> behind Kamailio Dispatcher, I've attached a call from extension 1001 to<br>
>>>> 1002, which fails. It just hangs for some time and then says that 1002<br>
>>>> cannot be found, and then the voicemail for it comes up.<br>
>>>> 2011/11/13 Henrik Aagaard Sørensen <<a href="mailto:henrikaagaardsorensen@gmail.com">henrikaagaardsorensen@gmail.com</a>><br>
>>>>><br>
>>>>> I'm trying to get the setup Kamailio Dispatcher -> FreeSwitch to work.<br>
>>>>> I've setup Kamailio via<br>
>>>>> this: <a href="http://www.kamailio.org/docs/modules/stable/modules_k/dispatcher.html" target="_blank">http://www.kamailio.org/docs/modules/stable/modules_k/dispatcher.html</a><br>
>>>>> I've installed FreeSwitch from scratch on Ubuntu<br>
>>>>> via: <a href="http://wiki.freeswitch.org/wiki/Ubuntu_Quick_Start" target="_blank">http://wiki.freeswitch.org/wiki/Ubuntu_Quick_Start</a><br>
>>>>> Now, when registering extensions via Kamailio Dispatcher I'm able to<br>
>>>>> call to FreeSwitch and listen to hold music. But that's it. I'm not able to<br>
>>>>> call between extensions etc.<br>
>>>>> Can anyone help me setting up FreeSwitch to accept registration, calls<br>
>>>>> etc. from Kamailio and everything else that is needed to use FreeSwitch<br>
>>>>> behind a load balancer?<br>
>>>>> I'm very new to FreeSwitch, but I'm trying to use the terminal (without<br>
>>>>> any GUI etc.) as I want the installation to be as clean as possible. So I<br>
>>>>> would prefer very precise help, as I'm still getting hold of FreeSwitch.<br>
>>>><br>
>>>><br>
>>>> _________________________________________________________________________<br>
>>>> Professional FreeSWITCH Consulting Services:<br>
>>>> <a href="mailto:consulting@freeswitch.org">consulting@freeswitch.org</a><br>
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>>>><br>
>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server<br>
>>>> <a href="http://www.cudatel.com" target="_blank">http://www.cudatel.com</a><br>
>>>><br>
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>>>><br>
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>>>><br>
>>><br>
>>><br>
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>>><br>
>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server<br>
>>> <a href="http://www.cudatel.com" target="_blank">http://www.cudatel.com</a><br>
>>><br>
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>>><br>
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>>><br>
>><br>
>><br>
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>><br>
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>><br>
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>><br>
><br>
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><br>
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</div></div></blockquote></div><br>