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.