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").