I am already have some practice to integrate Kamailio with Asterisk, when all users creates and registers in Kamailio, and calls go to/from Asterisk with static "host=kamailio_ip" settings for each user on Asterisk side.
I can't (don't know - how to) use in same way integration with FreeSWITCH. Can't create in FS directory structure a user with "host=kamailio_ip", FS require registration.
Maybe I can register user on Kamailio and send additional registration request to FS with src ip changed to kamailio (lan ip)?
P.S. Reading FreeSWITCH 1.2 book in progress...
On Mar 28, 2014, at 11:36 AM, Alexandr Usov blessendor@gmail.com wrote:
I am already have some practice to integrate Kamailio with Asterisk, when all users creates and registers in Kamailio, and calls go to/from Asterisk with static "host=kamailio_ip" settings for each user on Asterisk side.
I can't (don't know - how to) use in same way integration with FreeSWITCH. Can't create in FS directory structure a user with "host=kamailio_ip", FS require registration.
Freeswitch does not require registration. What are you trying to use freeswitch for? Voicemail, B2BUA (transcoding), ivr? See the acl configuration or don't use the directory at all.
HTH --FC
2014-03-28 18:16 GMT+02:00 Frank Carmickle frank@carmickle.com:
Freeswitch does not require registration. What are you trying to use freeswitch for? Voicemail, B2BUA (transcoding), ivr? See the acl configuration or don't use the directory at all.
I want use Kamailio only for registrations and use rtpproxy for registered peers. So all PBX features I want to use on FS (and/or Asterisk).
So acl style can't serve Voicemail and presence features for not registered (on FS) users?
On 31/03/14 13:07, Alexandr Usov wrote:
2014-03-28 18:16 GMT+02:00 Frank Carmickle <frank@carmickle.com mailto:frank@carmickle.com>:
Freeswitch does not require registration. What are you trying to use freeswitch for? Voicemail, B2BUA (transcoding), ivr? See the acl configuration or don't use the directory at all.I want use Kamailio only for registrations and use rtpproxy for registered peers. So all PBX features I want to use on FS (and/or Asterisk).
So acl style can't serve Voicemail and presence features for not registered (on FS) users?
For presence you don't need registration. For example, for MWI just forward the subscribe request to freeswitch, being sure it matches the voicebox/voicemail id.
The default config for kamailio has logic for forwarding mwi subscriptions to voicemail server.
Cheers, Daniel
On Mar 31, 2014, at 7:22 AM, Daniel-Constantin Mierla miconda@gmail.com wrote:
On 31/03/14 13:07, Alexandr Usov wrote:
2014-03-28 18:16 GMT+02:00 Frank Carmickle frank@carmickle.com:
Freeswitch does not require registration. What are you trying to use freeswitch for? Voicemail, B2BUA (transcoding), ivr? See the acl configuration or don't use the directory at all.
I want use Kamailio only for registrations and use rtpproxy for registered peers. So all PBX features I want to use on FS (and/or Asterisk).
So acl style can't serve Voicemail and presence features for not registered (on FS) users?
For presence you don't need registration. For example, for MWI just forward the subscribe request to freeswitch, being sure it matches the voicebox/voicemail id.
The default config for kamailio has logic for forwarding mwi subscriptions to voicemail server.
You would need to create the voicemail boxes in the voicemail database for each user. You should do this when creating the user.
--FC
Hello,
On 31/03/14 14:10, Frank Carmickle wrote:
On Mar 31, 2014, at 7:22 AM, Daniel-Constantin Mierla miconda@gmail.com wrote:
On 31/03/14 13:07, Alexandr Usov wrote:
2014-03-28 18:16 GMT+02:00 Frank Carmickle frank@carmickle.com:
Freeswitch does not require registration. What are you trying to use freeswitch for? Voicemail, B2BUA (transcoding), ivr? See the acl configuration or don't use the directory at all.
I want use Kamailio only for registrations and use rtpproxy for registered peers. So all PBX features I want to use on FS (and/or Asterisk).
So acl style can't serve Voicemail and presence features for not registered (on FS) users?
For presence you don't need registration. For example, for MWI just forward the subscribe request to freeswitch, being sure it matches the voicebox/voicemail id.
The default config for kamailio has logic for forwarding mwi subscriptions to voicemail server.
You would need to create the voicemail boxes in the voicemail database for each user. You should do this when creating the user.
they can be generated on the fly as well -- a starting point here:
- http://kb.asipto.com/freeswitch:kamailio-3.3.x-freeswitch-1.2.x-sbc#user_dir...
Cheers, Daniel
Hi Alexandr
On 28 March 2014 15:36, Alexandr Usov blessendor@gmail.com wrote:
I am already have some practice to integrate Kamailio with Asterisk, when all users creates and registers in Kamailio, and calls go to/from Asterisk with static "host=kamailio_ip" settings for each user on Asterisk side.
I can't (don't know - how to) use in same way integration with FreeSWITCH.
You could just set <param name="outbound-proxy" value="kamailio_ip"/> in the SIP profile which would send ALL calls via Kamailio.
Can't create in FS directory structure a user with "host=kamailio_ip", FS require registration.
For each user add:
<param name="dial-string" value="sofia/internal/${dialed_user}@kamailio_ip "/>
This replaces the default "dial-string". See more details here: http://wiki.freeswitch.org/wiki/XML_User_Directory_Guide#Dial_String
Maybe I can register user on Kamailio and send additional registration request to FS with src ip changed to kamailio (lan ip)?
This can also work. Both Kamailio and FS support RFC 3327 and the Path header which can we used to tell FS to send calls for the user via Kamailio.
Note these are 3 separate solutions, you should probably only do one.
Regards, Richard
Thanks Richard, I will trying to use your solution on practice. Need roll-back of default FS configs, because it seems that my /dev/hands not working good)
2014-03-28 20:48 GMT+02:00 Richard Brady rnbrady@gmail.com:
Hi Alexandr
On 28 March 2014 15:36, Alexandr Usov blessendor@gmail.com wrote:
I am already have some practice to integrate Kamailio with Asterisk, when all users creates and registers in Kamailio, and calls go to/from Asterisk with static "host=kamailio_ip" settings for each user on Asterisk side.
I can't (don't know - how to) use in same way integration with FreeSWITCH.
You could just set <param name="outbound-proxy" value="kamailio_ip"/> in the SIP profile which would send ALL calls via Kamailio.
Can't create in FS directory structure a user with "host=kamailio_ip", FS require registration.
For each user add:
<param name="dial-string" value="sofia/internal/${dialed_user}@kamailio_ip"/>
This replaces the default "dial-string". See more details here: http://wiki.freeswitch.org/wiki/XML_User_Directory_Guide#Dial_String
Maybe I can register user on Kamailio and send additional registration request to FS with src ip changed to kamailio (lan ip)?
This can also work. Both Kamailio and FS support RFC 3327 and the Path header which can we used to tell FS to send calls for the user via Kamailio.
Note these are 3 separate solutions, you should probably only do one.
Regards, Richard
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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