Re: [SR-Users] [SR-USERS] Kamailio issues with Asterisk Voicemail

[Daniel Tryba]

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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Re: [SR-Users] [SR-USERS] Kamailio issues with Asterisk Voicemail

[tegamjg]

El 05/07/2016 11:36, Daniel Tryba  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.
>
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Re: [SR-Users] [SR-USERS] Kamailio issues with Asterisk Voicemail

[Daniel Tryba]

On Mon, Jul 04, 2016 at 07:39:14PM -0400, TEG AMJG wrote:
> Now to the real question, is there anyway to include Asterisk voicemail
> functionality with the solution I already mentioned? the question is
> because, as I far as I know, Asterisk needs to associate mailboxes to its
> users. So i believe that i have to practically change most of my solution
> and find a way to make Kamailio register and locate users with the Realtime
> integration of Asterisk and map the registration accordingly to the table
> fields.
> 
> But it destroys most of my NAT and register solution since i also
> configured one of the private IPs as a peer in Asterisk sort of like in
> that guide

My solutions are similar to yours. The way I use asterisk voicemail is
to have a script sync kamailio users that want voicemail to the asterisk
realtime voicemail table and simply forward calls to voicemail to the
asterisk server as $ru="rvm-username@voicemail.instance" to access
voicemail and $ru="tvm-username@voicemail.instance" to leave voicemail
(via a failure route after timeout (t_set_fr()).


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[SR-Users] [SR-USERS] Kamailio issues with Asterisk Voicemail

[TEG AMJG]

Dear list:

I am quite new to Kamailio and i am really struggling to get some answer to
some questions i have been trying to get answers by reading documentation
and other questions or solutions around internet.

Even though i am integrating Kamailio with two Asterisk boxes with the
module dispatcher, i didnt follow the quite popular guide of Asterisk
Kamailio Realtime Integration (
http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb).
The reason why i didnt is that it actually behaved weird since i wanted
Kamailio to be a stateful proxy and also wanted Kamailio to act as a
Registrar Server and Location Server using subscriber and location tables
of kamailio database, keeping both Asterisk strictly as Media Servers.

Also i have been able to solve some NAT Traversal issues using RTPProxy and
not using those Asterisk boxes with their nat or qualify parameters to
solve NAT issues. Most of my solution started following this guide (
http://saevolgo.blogspot.com/2013/08/rtpproxy-revisited-kamailio-40.html)
which let me handle registration and location with Kamailio and also
changed it putting kamailio behind NAT.

Now to the real question, is there anyway to include Asterisk voicemail
functionality with the solution I already mentioned? the question is
because, as I far as I know, Asterisk needs to associate mailboxes to its
users. So i believe that i have to practically change most of my solution
and find a way to make Kamailio register and locate users with the Realtime
integration of Asterisk and map the registration accordingly to the table
fields.

But it destroys most of my NAT and register solution since i also
configured one of the private IPs as a peer in Asterisk sort of like in
that guide

[Kamailio]

type=friend

host=192.168.1.244

port=5060

disallow=all

allow=gsm

allow=g729

allow=alaw

allow=ulaw

Sorry for making the question so long and to be quite ignorant with some of
the concepts involving Kamailio with Asterisk


Thanks


Alejandro
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