[Kamailio-Users] Thomson ST2030 SIP contact problem
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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