[Kamailio-Users] Thomson ST2030 SIP contact problem
klaus.mailinglists at pernau.at
Mon Dec 1 13:28:12 CET 2008
The INVITE sent from Kamailio to Thomson phone does not trigger any
response. There are various possible reasons:
1. INVITE is ignored by Thomson phone
2. INVITE does not make it thorugh to the Thomson phone
2.1 either sent to the wrong port
2.2 or the NAT binding time out, thus NAT does not forward correctly
Thus, verify if the INVITE is received by the NAT device and forwarded
to the Thomson phone (e.g. putting a hub between the NAT router and the
phone). REGISTER with the Thomson phone and then immediately after call
it (linksys->thomson) - this should work as the binding should be alive
just after the registration.
The problem could also be caused by a buggy NAT router or VPN client or
To further debug this issue you could also try the Thomson phone with
another VoIP service (e.g. iptel.org or ekiga.net) or try the Thomson
phone from another access (e.g. try it at home bypassing your company
You could also try to avoid port 5060, e.g. Put the proxy on port 5678
and also use other ports locally for the SIP clients. SIP ALGs
(application level gateways) usually are triggered by port 5060.
Samuel Muller schrieb:
> On Mon, Dec 1, 2008 at 12:24 PM, Klaus Darilion
> <klaus.mailinglists at pernau.at <mailto:klaus.mailinglists at pernau.at>> wrote:
> 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?
> oups, sorry, I mean : SIP messages are ok, there is all the sig process.
> the architecture is :
> linksys + thomson -> cisco 827 -> SDSL -> our backbone which have a
> firewall for VPN (so NAT and NAPT are applied here), then the kamailio
> with a public ip.
> you have the 100 trying, 180 ringing in the SIP message, but there's no
> ring-back tone for the callee.
> in the attached file :
> linksys to linksys, where all the call process is ok (sip + rtp)
> thomson to linksys, idem
> linksys to thomson, sip ok but rtp apparently not.
> I forgot the firewall for the vpn, rtp proxying is required, sorry - so
> yes rtp proxy must be used.
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