Re: [SR-Users] Kamailio Asterisk Followup
I route calls to Kamailio only if user is registered there. Other calls I route directly to outbound provider. On Wed, Feb 15, 2012 at 7:45 PM, Olle E. Johansson o...@edvina.net wrote: 15 feb 2012 kl. 18:29 skrev Stoyan Mihaylov: 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. Don't forget to set outbound proxy in asterisk, so all calls, regardless of destination, go to Kamailio. /O ___ 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
Re: [SR-Users] Kamailio Asterisk Followup
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 ___ 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
Re: [SR-Users] Kamailio Asterisk Followup
15 feb 2012 kl. 18:29 skrev Stoyan Mihaylov: 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. Don't forget to set outbound proxy in asterisk, so all calls, regardless of destination, go to Kamailio. /O ___ 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
[SR-Users] Kamailio Asterisk Followup
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
