[Kamailio-Users] Thomson ST2030 SIP contact problem

Klaus Darilion klaus.mailinglists at pernau.at
Mon Dec 1 12:30:10 CET 2008

Samuel Muller schrieb:
 > Hey Klaus,
 > first, some answers :
 > ->  when a thomson is the callee, there's no ringing even if 
indicated into the SIP message.
 > -> when a thomson is the caller, no problem, there's a ring, and the 
call is ok with audio.

Please be a bit more specific: What does "no ring" mean?
  No "180 ringing" response from callee to caller?
  "180 ringing" response but no "ring-back" at the caller's client?

 > -> the SIP mesages are the same in the 2 cases.
 > incoming invite are exactly the same (but the source IP, and the 
"user=phone" parameter, for sure).
 > outgoing invite, in the 2 cases, replaces the source IP on the "c" 
line into the SDP by the RTP proxy IP (but at the end, it indicates 

a=nortpproxy:yes is fine. This line will signal to upstream proxies that 
an RTP proxy was already activated and there is no need to activate 
further RTP proxies.

Please post a complete ngrep trace (INVITE.....BYE) for the sucessfull 
and for the unsuccessful call.

 > Did I force 2 times the RTP proxy, or is there a


 > fix_nated_contact/fix_nated_register pbm ?

For REGISTER handling you use fix_nated_register. For all other cases 
you use fix_nated_contact.

BTW: What do you mean by "there's no RTP proxying" in the "not ok" case?


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