14 Feb
2012
14 Feb
'12
3:18 p.m.
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
15 Feb
15 Feb
2:36 a.m.
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(a)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(a)lists.sip-router.org
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12:29 p.m.
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(a)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(a)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(a)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(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
12:45 p.m.
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
16 Feb
16 Feb
8:07 a.m.
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(a)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(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
4698
days inactive
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4 comments
4 participants
participants (4)
-
Greg Mannie -
Mark Sayer -
Olle E. Johansson -
Stoyan Mihaylov