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
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 greg@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
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 datapipes@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 greg@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
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
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 oej@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
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Greg Mannie -
Mark Sayer -
Olle E. Johansson -
Stoyan Mihaylov