2 Feb
2004
2 Feb
'04
7:07 a.m.
Nothings behind NAT (I dont use NAT,.. me hate)
I havent done anything with Eatherreal yet. Just wonderd if anybody had
seen a simular problem
- Atle
on my SIP gateway, I have a statement :
if (uri=~"sip:8[0-9][3.5]*") {
strip(1);
rewritehostport("voip-vm.domain.com");
t_relay();
break;
};
that Im using to test my voicemail server
on the voicemail server Iv got this :
route{
# Voicemail specific configuration - begin
if(method=="ACK" || method=="INVITE" ||
method=="BYE"){
if(t_newtran()){
t_reply("10","Trying -- just wait a minute
!");
if(method=="INVITE"){
log("**************** vm start - begin
******************\n");
if(!vm("/tmp/am_fifo","voicemail")){
log("could not contact the answer
machine\n");
t_reply("500","could not
contact the answer machine");
};
log("**************** vm start - end
******************\n");
break;
};
if(method=="BYE"){
log("**************** vm end - begin
******************\n");
if(!vm("/tmp/am_fifo","bye")){
log("could not contact the answer
machine\n");
t_reply("500","could not
contact the answer machine");
};
log("**************** vm end - end
******************\n");
break;
};
}
else {
log("could not create new transaction\n");
sl_send_reply("500","could not create new
transaction");
};
};
# Voicemail specific configuration - end
(the of the stanard ser.cfg)
* Klaus Darilion <klaus.mailinglists(a)pernau.at> [040202 12:55]:
a little bit more information about your setup would
be useful! are the
clients behind NAT? have you analyzed the SIP message flow (e.g. using
ethereal)?
klaus
Atle Samuelsen wrote:
Hey again
Iv just got a weird problem.. When I connect my ip phone to the
voicemail gateway.. it works Perfect. I get the voicemail message and
everything.. Tho..
When I try to call via the other gateway (my regualar ser).. I get to
the voicemail.. it picks up the phone.. tho No voice back..
anybody had this problem before?
- Atle
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