Re: [Serusers] Re: [Users] rfc3264 and nathelper module - sr-users

29 Nov 2005

On Tue, 29 Nov 2005 14:46:55 +0100 (CET), harry gaillac wrote
...

 REGISTER:

         if (!is_from_local()) {
                 sl_send_reply("401", "Unauthorized");
                 break;
         };
                 rewritehostport("nxs.yi.org:5050");
                 t_relay_to_udp("nxs.yi.org","5050"); 

So... you want your users to register both with Asterisk AND with SER? What
for? Why not use some form of SER authorisation to control access to the
Asterisk server? I'm not sure what you're trying to do here...

...

 INVITE:
         rewritehostport("nxs.yi.org:5050"); 
What's the context around this? You want ALL calls to go straight to Asterisk? 

It just seems... I don't know... weird. I assume you're doing this because SER
handles presence and Asterisk doesn't... but everything else you want to be
controlled by Asterisk, including user registration, etc, etc.

For your invite, btw, you would need to also have a t_relay_to_udp, or
alternatively, you can do it as:

rewritehostport("nxs.yi.org:5050");
forward(uri:host,uri:port);
break;

...

 sip.conf:
 [general]
 context=local                   ; Default context for
 incoming calls
                                 ; if asterisk was
 compiled with OSP support.
 realm=nxs.yi.org                ; Realm for digest
 authentication
                                 ; defaults to
 "asterisk"
                                 ; Realms MUST be
 globally unique according to RFC 3261
                                 ; Set this to your
 host name or domain name
 bindport=5050                   ; UDP Port to bind to
 (SIP standard port is 5060)
 bindaddr=nxs.yi.org             ; IP address to bind
 to (0.0.0.0 binds to all)
 srvlookup=yes                   ; Enable DNS SRV
 lookups on outbound calls
 tos=lowdelay                    ;
 lowdelay,throughput,reliability,mincost,none
 maxexpirey=3600                 ; Max length of
 incoming registration we allow
 defaultexpirey=1000             ; Default length of
 incoming/outoing registration
 allow=all                       ; First disallow all
 codecs
 musicclass=default              ; Sets the default
 music on hold class for all SIP calls
 language=fr                     ; Default language
 setting for all users/peers
 rtptimeout=60                   ; Terminate call if 60
 seconds of no RTP activity
 tpholdtimeout=300               ; Terminate call if
 300 seconds of no RTP activity
 useragent=Asterisk PBX          ; Allows you to change
 the user agent string
 dtmfmode = rfc2833              ; Set default dtmfmode
 for sending DTMF. Default: rfc2833
 promiscredir = no               ; If yes, allows 302
 or REDIR to non-local SIP address

 [84]
 type=friend
 secret=84
 username=84
 context=local
 host=dynamic
 mailbox=84
 allow=all

 [85]
 type=friend
 secret=85
 username=85
 context=local
 host=dynamic
 mailbox=85
 allow=all

In your blocks there, you might want to add
insecure=very
to allow registered hosts to call without reauthenticating -- just to speed
things up a little (optional)

one IMPORTANT thing you want to add to each block just to make sure it's there is:

canreinvite=yes

This allows the RTP traffic to bypass asterisk and just go from UA to UA
without taking up bandwidth.  It will NOT, however, work well for NATted users
without playing around with rewriting the SDP (using something like a
nathelper fix_nated_sdp("3") call)

Rereading your original emails, I now think I understand that this is what you
were asking about before (correct me if I'm wrong)... was the fix_nated_sdp
not working correctly in some way?

N.