Displaying 5 results from an estimated 5 matches for "l_info".
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2005 May 09
1
Asterisk + SER and NAT
...;Too Many Hops");
break;
};
if ( msg:len > max_len ) {
sl_send_reply("513", "Message too big");
break;
};
# if(method != "REGISTER")
# {
# xlog("L_INFO", "Got a %rm (URI = %ru) from %is");
# };
# By default we are already responding to the learned IP:PORT that
the message came from.
# This will properly fix the contact header. Asterisk will fix
the SDP (actually it learns it).
force_rport();...
2007 May 12
3
Asterisk High-Capacity Stability
...er_proxies WHERE
> customer_id = $avp(S:customer_id) AND active = true",
> "$avp(S:proxy_ip);$avp(S:proxy_port)");
>
> if(!is_avp_set("$avp(S:proxy_ip)") ||
> !is_avp_set("$avp(S:proxy_port)")) {
> xlog("L_INFO", "target-das - [$ci] - Active proxy not
> found.\n")
> ;
> sl_send_reply("404", "Not Found");
> exit;
> }
>
> xlog("L_INFO", "target-das - [$ci] - Resolved proxy
> $avp(S:p...
2009 Apr 13
0
opensips and asterisk canreinvite
Hi,
I'm using opensips as the registrar server for my users.
I am redirecting calls going out to pstn to my asterisk server.
call flow is basically:
ua --> opensips server --> * server --> sip gateway provider
if (uri=~"sip:00[0-9]*@sip\.myserver\.com") {
xlog("L_INFO", "Call to PSTN\n");
#strip(2);
#prefix("011");
rewritehostport("20.21.22.23:6050"); <--- IP and Port of * Server
route(1);
exit;
}
call routing works properly, but i would like for the rtp not to go thru
asterisk, i'm using the canrei...
2014 Apr 25
3
Proper way to make Asterisk recognize SIP trunk of incoming INVITE when IP is not available
I am currently preparing a kamailio-asterisk combination. The asterisk installation uses realtime for SIP. The kamailio configuration was based on the reference at http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb but has been
heavily modified. Currently asterisk runs on localhost and only listens on SIP/RTP at 127.0.0.1 . Therefore, all of the SIP traffic appears to
2004 Jan 20
1
Toll-Free Gateway Beta Test: freenum.org
...xlog("L_ERR", "%is [%Tf]: %rm %fu -> %ru [R4]: freenum.org destination\n");
break;
};
#end
# nothing found, try PSTN
if (method=="CANCEL" || method=="BYE" || method=="ACK") {
xlog("L_INFO", "%is [%Tf]: %rm %fu -> %ru [R4]: just forwarding to PSTN");
rewritehostport("1.2.3.4:5060");
t_relay();
break;
};
====== end of configurations ========