[SR-Users] home pbx server experience

[Moacir Ferreira]

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