15 Nov
2005
15 Nov
'05
11:14 a.m.
sounds to me like you have a looping problem in your script. I had
something similar when using the example from OnSIP.org. The
loose_route bit needed to be inside a condition:
if (uri!=myself){
if (loose_route()) {
route(1);
};
};
xlog/ngrep is your friend here as you will be able to see which message
is being sent between the two servers.
Noel
Giovanni Balasso wrote:
Alle 09:47, mercoledì 02 novembre 2005, Matteo Piazza
ha scritto:
I have Ser and asterisk on the same machine.
When i try to call with a SIP phone registred on asterisk another sip
phone also registred on asterisk through SER I receve this error message:
Too many hops
Too many hops is usually reached when there is no rule (or no way) to deliver
sip message, adding some log(), or better xlog(), to your routing script
could help you (and us) debugging and understanding what's wrong, and which
method(s) fail.
if (method == "INVITE") {
if (uri =~"^sip:0[0-9]*@*"){
log(1, "Check 1 succed Forwarding to Asterisk\n");
rewritehostport("192.168.9.97:5061");
t_relay();
break;
};
};
I don't think this will solve your problem but in my experience I had better
result with t_relay_to_udp("192.168.9.97","5061") than
rewritehostport("192.168.9.97:5061").