El 05/07/2016 11:36, Daniel Tryba <d.tr...@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.html > 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 _______________________________________________ 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
