<html><head></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; ">Hi,<div><br></div><div>I'm actually trying to configure the same thing right now - with opensips though, but (afaik) it uses the same dispatcher module.</div><div><br></div><div>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.</div><div><br></div><div>To fix that, I added record routing in the proxy configuration so it stays in the path.</div><div><br></div><div>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.</div><div><br></div><div>To get calls originating from fs to go through the proxy before going to the client, I tried:</div><div><br></div><div>originate sofia/some_profile/sip:user@mydomain.com;fs_path=sip:ip_of_sip_proxy:5060 &park</div><div><br></div><div>which worked. Then I found out about the sip_route_uri variable that can be set in the bridge string:</div><div><br></div><div>originate {sip_route_uri=sip:ip_of_sip_proxy:5060}sofia/some_profile/sip:some_user@mydomain.com &park</div><div><br></div><div>which seemed to do the same thing (does anyone know if there's any difference between the two variants ?)</div><div><br></div><div>Then I tried putting the sip_route_uri in the dial-string param of the domain (or user), for example:</div><div><br></div><div><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})}"/></div><div><br></div><div>so now it's possible to just call the user as:</div><div><br></div><div>originate <a href="mailto:user/some_user@mydomain.com">user/some_user@mydomain.com</a> &park</div><div><br></div><div>nice !</div><div><br></div><div>And then I found out that you can just set an outbound-proxy param in the sip profile that's used for originating:</div><div><br></div><div><div><param name="outbound-proxy" value="<a href="sip:ip_of_sip_proxy:5060">sip:ip_of_sip_proxy:5060</a>"/></div></div><div><br></div><div>so now you can leave out the sip_route_uri or fs_path variables, it just works.</div><div><br></div><div>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.</div><div><br></div><div>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.</div><div><br></div><div>regards,</div><div><br></div><div>Leon</div><div><br></div><div><br></div><div><br></div><div><br><div><div>On Nov 14, 2011, at 6:40 AM, Henrik Aagaard Sørensen wrote:</div><br class="Apple-interchange-newline"><blockquote type="cite"><div bgcolor="#FFFFFF"><div>The handling of several FS are not a issue, actually easy enough.</div><div><br></div><div>I would like to have as little load on Kamailio as possible, as it just should load balance.</div>
<div><br></div><div>Also, having to handle users on both Kamailio and FS makes unecessary work 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></div><div></div><blockquote type="cite"><div>Hi again,<div><br></div><div>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. </div>
<div>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)</div>
<div><br></div><div>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.</div>
<div><br></div><div>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!</div><div><br></div>
<div>--</div><div>Regards,</div><div>Sammy<br><br><div class="gmail_quote">2011/11/14 Henrik Aagaard Sørensen <span dir="ltr"><<a href="mailto:henrikaagaardsorensen@gmail.com">henrikaagaardsorensen@gmail.com</a>></span><br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">Hi Sammy.<div><br></div><div>I've actually removed registration and presence from Kamailio, so all it does is dispatch everything to FreeSwitch.</div>
<div><br></div><div>Currently I only have 1 FreeSwitch, for testing this basic setup.</div>
<div>Next move would be 2 FreeSwitch with 1 common database etc. But that's later.</div><div><br></div><div>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.</div>
<div><br></div><div>But when going through Kamailio Dispatcher it fails between the extensions.</div><div><br></div><div>So I guess there should be some more setup in FreeSwitch when using a load balancer (dispatcher) in front of it.</div>
<div><br></div><div>My setup is just Kamailio -> FreeSwitch and FreeSwitch should handle everything.<div><div class="h5"><br><br><div class="gmail_quote">On Mon, Nov 14, 2011 at 5:56 AM, Sammy Govind <span dir="ltr"><<a href="mailto:govoiper@gmail.com" target="_blank">govoiper@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Hi,<div><br></div><div>If everything is setup as you expected it then don't read this > <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>,</div>
<div>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.</div><div><br></div><div>Once that is implemented then apply dispatcher module on the route which says [REGFWD] or [TOASTERISK].</div>
<div><br></div><div>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.</div><div><br></div><div>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?</div>
<div>--</div><div>Regards,</div><div>Sammy</div><div><br></div><div><br><div class="gmail_quote"><div><div>2011/11/14 Henrik Aagaard Sørensen <span dir="ltr"><<a href="mailto:henrikaagaardsorensen@gmail.com" target="_blank">henrikaagaardsorensen@gmail.com</a>></span><br>
</div></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><div>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.<div>
<div><div>
<br><div class="gmail_quote">2011/11/13 Henrik Aagaard Sørensen <span dir="ltr"><<a href="mailto:henrikaagaardsorensen@gmail.com" target="_blank">henrikaagaardsorensen@gmail.com</a>></span><br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
I'm trying to get the setup Kamailio Dispatcher -> FreeSwitch to work.<div><br></div><div>I've setup Kamailio via 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></div>
<div><br></div><div>I've installed FreeSwitch from scratch on Ubuntu via: <a href="http://wiki.freeswitch.org/wiki/Ubuntu_Quick_Start" target="_blank">http://wiki.freeswitch.org/wiki/Ubuntu_Quick_Start</a></div><div>
<br></div><div>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.</div>
<div><br></div><div>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?</div><div><br></div><div>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.</div>
</blockquote></div><br></div>
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