I did - registration is purely in Kamailio. In Asterisk - I created sip account for Kamailio based on IP address without username and password. This way - all calls from Kamailio go to Asterisk without problems. In Kamailio I allowed calls from Asterisks. You do not need realtime in Asterisk, because Kamailio do all registrations perfectly well.
On Wed, Feb 15, 2012 at 9:36 AM, Mark Sayer <datapi...@avtb.co.nz> wrote: > We've created custom processes to accomplish this. FreePBX may work > but as it will assume a standalone Asterisk setup it may just cause > problems. > > Mark > > On Wed, Feb 15, 2012 at 7:18 AM, Greg Mannie <g...@latigi.com> wrote: > > Hello > > > > Sorry for another newbie question, but eventually with your "greatly > > appreciated" help I will get proficient in this application. > > > > After reading much RFC reading and docs for Kamailio I see where the > > benefits of using the sip proxy for registering devices while using > asterisk > > for voicemail or ivr etc. has great benefit. > > > > I am not finding much on my end user interaction. If I use realtime > > integration and have multi domain use on kamailio, how do I allow the end > > user to configure their own IVR? Is it possible to use modules like the > > call flow control from freepbx and allow users to configure this > themselves? > > > > Regards, > > > > Greg > > > > > > > > > > > > _______________________________________________ > > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > > sr-users@lists.sip-router.org > > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users@lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >
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