[asterisk-users] DUNDi + SIP Realtime

[Vinícius Fontes]

Good afternoon gentlemen (and ladies).

A costumer of mine has many servers and each one maps their SIP extensions to the others via DUNDi. It works like a charm. SIP extensions can only register at one server, the one they "belong" to. In case one extension wants to call other that is registered in another server, DUNDi takes care of that by calling the other server using IAX2 and G.729 codec. It's important to clarify that each server today works as a completely independent PBX, talking to each other using IAX2 that is routed via a MPLS network. Also, the servers are very distant from each other physically.

Today the extensions are being mapped in DUNDi like this:

[dundi-internal]
exten => _70XX,1,Noop()

And all extensions configurations are done in sip.conf. No realtime is being used, yet.




Now the customer wants to take a step further and make it possible that *every* SIP extension could be able to register in *every* server. That would make possible for them to use DNS to automatically "find" the closest PBX and make the extension register on that one.

So far I considered the following for this project:

- Moving all SIP extensions from individual sip.confs to one MySQL database, and point all servers to that one
- Configure sip.conf on each machine like this:

regcontext=dundi-internal
rtcachefriends=yes
rtsavesysname=yes
rtupdate=no
rtautoclear=yes
ignoreregexpire=no

That way each time an extension registers, Asterisk would add an extension to the dundi-internal context, which as you guessed, is the one being mapped to the other servers. So instead of mapping extensions using wildcards, the extensions will be mapped individually.

extensions.conf would be something like this:

[internal]
;Tries to make the call using SIP, in the case
;the extension is registered in this server
;If it's not, switches to DUNDi
exten => _XXXX,1,Dial(SIP/${EXTEN},60)
exten => _XXXX,n,NoOp(DIALSTATUS = ${DIALSTATUS})
exten => _XXXX,n,NoOp(FROM_DUNDI = ${FROM_DUNDI})
exten => _XXXX,n,GotoIf($["${FROM_DUNDI}" = "1"]?end:start)
exten => _XXXX,n(start),Answer()
exten => _XXXX,n,Playback(vm-dialout)
exten => _XXXX,n,Goto(dundi-internal-helper,${EXTEN},1)
exten => _XXXX,n(end),Noop(Loop detected. Hanging up.)
exten => _XXXX,n,Hangup()

[dundi-internal-helper]
switch => DUNDi/dundi_internal

[from-dundi]
exten => _XXXX,1,Set(FROM_DUNDI=1)
exten => _XXXX,n,Dial(SIP/${EXTEN},60)


So far it's working fine in a test lab with 2 servers running Asterisk 1.6.0.15.

For the gurus out there: is there something that I'm doing terribly wrong, that would break everything and make the universe collapse into itself when I apply the same principle on production?

I'll be happy to provide more details in case there are any doubts. I really appreciate your feedback, no matter what is it. :)



Vinícius Fontes
www.asteriskforum.com.br - Informações e discussão sobre Asterisk e telefonia IP