[SR-Users] Help with Asterisk RT integration

[Daniel Tryba]

On Thursday 13 December 2012 12:09:46 Daniel-Constantin Mierla wrote:
> > The problem I'm seeing currently is that when a call is passed down a SIP
> > trunk to an end user on the (K) platform we're losing the DNID
> > 
> > Asterisk delivers the call to SIP/account/DNID
> > 
> > (K) however just tries to deliver the call to DNID at domain which comes
> > back as not found.
> 
> Maybe you can post ngrep trace of such call, taken on kamailio server, 
> to see the incoming and outgoing traffic. I didn't understand what is 
> the real problem, but with the ngrep trace we can see the signaling and 
> tell where it may be the issue.

I'm not using RT integration but I see the same "problem"[1] if I interpret 
Jon's message correctly.

The asterisk to kamailio invite will be something like:
INVITE sip:31880100781 at kamailio SIP/2.0
From: "Daniel" <sip:0402938661 at asterisk>;tag=as67f9e721
To: <sip:31880100781 at kamailio>

If you are using alias_db_lookup on invite to rewrite the request to a 
username at the kamailio domain to forward the request to a registered local 
subscriber the DNID will be replaced by the username:

INVITE sip:username at endpoint:5060;transport=udp SIP/2.0
From: "Daniel" <sip:0402938661 at asterisk>;tag=as67f9e721
To: <sip:31880100781 at kamailio>

Essentially the difference between asterisk extensions:
exten => 123,n,Dial(SIP/123 at username)
and
exten => 123,n,Dial(SIP/username)

A solution might be to forward to and external domain with unchanged 
"username", but that doesn't work if the current setup is natted without 
portforwards. 

It shouldn't be to hard to write a custom sql query to figure out to which 
subscriber a block of number belongs and get the ip/port/protocol combinations 
from the subscriber list to create branches to t_relay() the INVITE. 

[1]: I'm undecided whether if this is a problem of feature. Aastra devices 
can't handle the DNID format, but I prefer DNID INVITES for PBXs. But a decent 
PBX should be able to use the To header to figure out the dialed extension 
(but often not without manual reconfiguration).

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