El 05/07/2016 11:36, Daniel Tryba d.tryba@pocos.nl escribió:
Please keep the mailinglist in the loop, so everybody might benefit from our ramblings :)
Still there are few things i dont understand, i am not using asterisk just as a voicemail server since they are actually handling also the calls passing first from kamailio and being load balanced to those asterisk boxes. May i still use call forwarding as you are using it? (Both asterisk have a shared storage with a clustered filesystem, so both will be able to see voice messages)
Yes I think so. I use a seperate machine for voicemail but I see no problem with other uses (I used to use it for playback of messages and transcoding ebtween incompatible endpoints).
By using the prefixes in kamailio to the username in $ru I have in the extensions.conf:
exten => _tovm-.,1,NoOp(leave voicemail) exten => _tovm-.,n,Answer() exten => _tovm-.,n,Set(CHANNEL(language)=nl) exten => _tovm-.,n,Voicemail(${EXTEN:5},us) exten => _tovm-.,n,Playback(Goodbye) exten => _tovm-.,n,Hangup()
exten => _getvm-.,1,NoOp(read voicemail) exten => _getvm-.,n,Set(CHANNEL(language)=nl) exten => _getvm-.,n,VoicemailMain(${EXTEN:6}) exten => _getvm-.,n,Hangup()
The other question is that i actually though that you need asterisk to have users configured in sipusers realtime table to associate their mailboxes, which i dont have since those users are stored in the subscriber table of kamailio. So am i still able to configure voicemail like you are doing it by syncing with the voicemail table?, i really hope so haha
I forgot that fact. So yes I have a realtime sip users list (with host=dynamic,type=friend,insecure=port,invite, name/mailbox the kamailio username, no password (this machine is not directly accessible from outside))
Sorry, I think i rushed the last answer but if you could answer that one would be nice
How are you handling the calls? Just with kamailio/rtpproxy? Because i am also using asterisk for calls with dial application and for nat issues (with kamailio behind nat) i am using also kamailio/rtpproxy for outside. All this with just handling users (registration and location) in the subscribe and location table of kamailio.
That is why i am not using sipusers table of asterisk because of nat was behaving weird using it that way.
Could it be possible to use both tables without expecting a different behaviour? Or is not, in the end, a good idea and i need to keep users in sipusers table?
You might not be able to have endpoints able to subscribe to notifications due to this. I baked something inspired by: http://saevolgo.blogspot.nl/2012/07/asterisk-behind-kamailio-voicemail-mwi.h... that appears to work for me.
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
On Tue, Jul 05, 2016 at 01:06:31PM -0400, tegamjg wrote:
The other question is that i actually though that you need asterisk to have users configured in sipusers realtime table to associate their mailboxes, which i dont have since those users are stored in the subscriber table of kamailio. So am i still able to configure voicemail like you are doing it by syncing with the voicemail table?, i really hope so haha
I forgot that fact. So yes I have a realtime sip users list (with host=dynamic,type=friend,insecure=port,invite, name/mailbox the kamailio username, no password (this machine is not directly accessible from outside))
Sorry, I think i rushed the last answer but if you could answer that one would be nice
How are you handling the calls? Just with kamailio/rtpproxy?
Kamailio as loadbalancer/registrar server with rtpengine (always).
UAC <-> loadbalancers (kam.) <-> registrars (kam.) <-> voicemail (ast.) <-------> rtpengine <------>
Because i am also using asterisk for calls with dial application and for nat issues (with kamailio behind nat) i am using also kamailio/rtpproxy for outside. All this with just handling users (registration and location) in the subscribe and location table of kamailio.
That is why i am not using sipusers table of asterisk because of nat was behaving weird using it that way.
Could it be possible to use both tables without expecting a different behaviour? Or is not, in the end, a good idea and i need to keep users in sipusers table?
Don't know, you have to try to see what happens. Difference in my asterisk setup is that I don't define the kamailio machines as peers except for the redirection servers (outbound only), everybody get access as "guest" and use dialplan logic instead. Defining people that want voicemail as users might screw with your peer for kamailio.
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Daniel Tryba -
tegamjg