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@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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