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<body class='hmmessage'><div dir='ltr'>Same problem here... I like Kamailio for the same reasons you posted. However, the UAC module is not for "dummies" like I am and also CSeq does not increased during authentication (<a href="http://kamailio.org/docs/modules/1.5.x/uac.html">http://kamailio.org/docs/modules/1.5.x/uac.html</a>), so the response may be rejected.<BR> <BR>I did a try on SEMS (<a href="http://www.iptel.org/sems">http://www.iptel.org/sems</a>). Although it looks easy to setup, it takes a while before we get the software "working logic". But I haven't having that much time to play with it so it may be my fault.<BR> <BR>I would be more than glad to work together (cooperate) with anyone willing to figure out an easy way to get a light, but still full UAC, working with Kamailio, were we would not be looking to setup a system to address 100.000 subscribers but home offices/small business. But it is more like that many people prefer Asterisk for this. And the main question is: Should we follow the trend and work with Asterisk? How much does Asterisk, being used as just a UAC, impact a small PC when running along with Kamailio? And yes, my key concern is not only voice but also video, what is a kind of cumbersome in Asterisk.<BR> <BR>Cheers!<BR> <BR>Moacir<br> <BR><div>> Date: Tue, 14 May 2013 20:40:47 -0300<br>> From: 4lists@gmail.com<br>> To: sr-users@lists.sip-router.org<br>> CC: ueberwachungsstaat@googlemail.com<br>> Subject: Re: [SR-Users] home pbx server experience<br>> <br>> Hi...<br>> <br>> Use UAC module to manage registrations and play a little with the config <br>> (INVITE section) to forward output calls correctly.<br>> ---<br>> Edson.<br>> <br>> Em 14/05/2013 11:27, u escreveu:<br>> > I would like to share my experience with kamailio and other home pbx servers.<br>> ><br>> > Kamailio on my kirkwood home router for my 6 SIP users is perhaps<br>> > overkill: I don't really need mysql and "scalability". But at last I<br>> > finally managed to make calling between registered users work stable.<br>> > My voip clients only work in all NAT scenarios if I work around some<br>> > bugs: to use csipsimple on android I had to change rtpproxy_manage()<br>> > to rtpproxy_manage("c") in kamailio's default config, so that problems<br>> > with conflicting c: entries in the SDP go away.<br>> ><br>> > I propose kamailio could ship with a special example<br>> > kamailio-compatible.cfg that doesn't try to be RFC compliant, but<br>> > compatible to the most common voip clients. Right now the only thing I<br>> > would change for this is the option for rtpproxy_manage, but I'm sure<br>> > others will know more common quirks that could safely be enabled to<br>> > increase compatibility. I think this compatibility idea is what yate<br>> > sticks to for their defaults. In freeswitch you also have to do it all<br>> > manually, and it's much more work to figure things out in their<br>> > enormous config files.<br>> ><br>> > The other SIP proxies I had tried before kamailio officially fit all<br>> > my requirements, including support for multihomed dynamic IPs, but<br>> > contrary to their claims it didn't work.<br>> > Yate was easy to set up, but the default dialplan is more confusing<br>> > than powerful and after having made everything work I realised yate<br>> > was clogging my CPU and RAM and after some time always randomly<br>> > stopped working. This is with only 2 users connected! It also wasn't<br>> > possible to fix NAT sdp while leaving the codecs section in the SDP<br>> > alone at the same time. I tried to debug the code, but the C++ was so<br>> > complex that I had to give up.<br>> > Freeswitch was much more difficult to setup, a multihomed setup with<br>> > dynamic IP was super buggy and it also didn't help that the<br>> > unintuitive configuration is all in complex unreadable XML<br>> > configuration files.<br>> ><br>> > Kamailio and rtpproxy don't officially support dynamic IP address, but<br>> > I can just restart both each time my DSL provider forces me to a new<br>> > IP address. This happens automatically in the night and is no big<br>> > hassle really. The most simple, least-featureful solution works best<br>> > it seems.<br>> ><br>> > Now the last problem I have with kamailio: I don't know how to connect<br>> > my accounts to my sip providers (i.e. Sipgate, Betamax, Dellmont).<br>> > I would like a simple way to do this, preferably without other<br>> > features that always seem to complicate the matters. Is there<br>> > something more lightweight and simple than asterisk, freeswitch and<br>> > yate, that people use successfully for this task together with<br>> > kamailio and rtpproxy?<br>> ><br>> > u<br>> ><br>> > _______________________________________________<br>> > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list<br>> > sr-users@lists.sip-router.org<br>> > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users<br>> > .<br>> ><br>> <br>> _______________________________________________<br>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list<br>> sr-users@lists.sip-router.org<br>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users<br></div> </div></body>
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