Hello,<br><br>I tried all the ways you told :<br><br>I moved the SIP phone at home, which I don't have any firewall and it does not pass through the entreprise fw.<br>so the SIP phone is directly connected to the proxy.<br>
<br>it registers well, no pbm, but the problem stay.<br>Impossible to make a call to the Thomson.<br><br>INVITE from any SIP phone (hard or soft, I tried with a Linksys, then a SJ Phone) through Kamailio is not going to the Thomson.<br>
All the others SIP stuff are working (Linksys to SJ Phone, ...).<br><br>I tried many configuration changes into the Thomson ST2030, unsuccessfully.<br>I mean it's not a NAT problem ...<br><br>Here you are the SIP messages in the kamailio debug from SJ Phone (0123451011) to the f***in' Thomson (0123451014) :<br>
<br>Dec 1 20:49:36 kamailio[29592]: -> incoming SIP buffer message: <br><br>INVITE <a href="mailto:sip%3A0123451014@sip.720.fr">sip:0123451014@sip.720.fr</a> SIP/2.0<br>Via: SIP/2.0/UDP <a href="http://192.168.1.3">192.168.1.3</a>;rport;branch=z9hG4bKc0a801030000004549343fcf39c16ac5000000f0<br>
Content-Length: 264<br>Contact: <<a href="http://sip:0123451011@192.168.1.3:5060">sip:0123451011@192.168.1.3:5060</a>><br>Call-ID: <a href="mailto:2E029B5E-1DD2-11B2-A585-993A960A9D75@192.168.1.3">2E029B5E-1DD2-11B2-A585-993A960A9D75@192.168.1.3</a><br>
Content-Type: application/sdp<br>CSeq: 2 INVITE<br>From: "sambook"<<a href="mailto:sip%3A0123451011@sip.720.fr">sip:0123451011@sip.720.fr</a>>;tag=409529589751851917<br>Max-Forwards: 70<br>To: <<a href="mailto:sip%3A0123451014@sip.720.fr">sip:0123451014@sip.720.fr</a>><br>
User-Agent: SJphone/1.60.299a/L (SJ Labs)<br>Proxy-Authorization: Digest username="0123451011",realm="<a href="http://sip.720.fr">sip.720.fr</a>",<br>nonce="493440fb0000001024641d0bca47789c4c6f68d81262f201",uri="<a href="mailto:sip%3A0123451014@sip.720.fr">sip:0123451014@sip.720.fr</a>",<br>
response="b8df3912d21ddd8aca40c0bf254bbdcf",cnonce="40952964931016109891",qop="auth",nc="00000001"<br><br>v=0<br>o=- 3437149775 3437149775 IN IP4 <a href="http://192.168.1.3">192.168.1.3</a><br>
s=SJphone<br>c=IN IP4 <a href="http://192.168.1.3">192.168.1.3</a><br>t=0 0<br>a=direction:active<br>m=audio 49168 RTP/AVP 0 8 3 101<br>a=rtpmap:0 PCMU/8000<br>a=rtpmap:8 PCMA/8000<br>a=rtpmap:3 GSM/8000<br><br><br>Dec 1 20:49:36 kamailio[29592]: -> outgoing SIP buffer message: <br>
<br>INVITE <a href="mailto:sip%3A0123451014@sip.720.fr">sip:0123451014@sip.720.fr</a> SIP/2.0<br>Via: SIP/2.0/UDP <a href="http://192.168.1.3">192.168.1.3</a>;rport;branch=z9hG4bKc0a801030000004549343fcf39c16ac5000000f0<br>
Content-Length: 264<br>Contact: <<a href="http://sip:0123451011@192.168.1.3:5060">sip:0123451011@192.168.1.3:5060</a>><br>Call-ID: <a href="mailto:2E029B5E-1DD2-11B2-A585-993A960A9D75@192.168.1.3">2E029B5E-1DD2-11B2-A585-993A960A9D75@192.168.1.3</a><br>
Content-Type: application/sdp<br>CSeq: 2 INVITE<br>From: "sambook"<<a href="mailto:sip%3A0123451011@sip.720.fr">sip:0123451011@sip.720.fr</a>>;tag=409529589751851917<br>Max-Forwards: 69<br>To: <<a href="mailto:sip%3A0123451014@sip.720.fr">sip:0123451014@sip.720.fr</a>><br>
User-Agent: SJphone/1.60.299a/L (SJ Labs)<br>Proxy-Authorization: Digest username="0123451011",realm="<a href="http://sip.720.fr">sip.720.fr</a>",<br>nonce="493440fb0000001024641d0bca47789c4c6f68d81262f201",uri="<a href="mailto:sip%3A0123451014@sip.720.fr">sip:0123451014@sip.720.fr</a>",<br>
response="b8df3912d21ddd8aca40c0bf254bbdcf",cnonce="40952964931016109891",qop="auth",nc="00000001"<br><br>v=0<br>o=- 3437149775 3437149775 IN IP4 <a href="http://192.168.1.3">192.168.1.3</a><br>
s=SJphone<br>c=IN IP4 <a href="http://192.168.1.3">192.168.1.3</a><br>t=0 0<br>a=direction:active<br>m=audio 49168 RTP/AVP 0 8 3 101<br>a=rtpmap:0 PCMU/8000<br>a=rtpmap:8 PCMA/8000<br>a=rtpmap:3 GSM/8000<br><br>In the attached file, the full kamailio debug level 9.<br>
<br>It seems that nothing is coming to the Thomson (I don't have any hub where I can sniff the frames).<br><br>"The truth is out there" ... :/<br><br><br>.desperate house Sam.<br><br><br><br><div class="gmail_quote">
On Mon, Dec 1, 2008 at 1:44 PM, Samuel Muller <span dir="ltr"><<a href="mailto:sml@720.fr">sml@720.fr</a>></span> wrote:<br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
many thanks Klaus,<br><br>I'll check tonight at home, and will reply to you after.<br><br>sincerely, thanks !<br><br>.Sam.<div><div></div><div class="Wj3C7c"><br><br><br><br><div class="gmail_quote">On Mon, Dec 1, 2008 at 1:28 PM, Klaus Darilion <span dir="ltr"><<a href="mailto:klaus.mailinglists@pernau.at" target="_blank">klaus.mailinglists@pernau.at</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">Hi Samuel!<br>
<br>
The INVITE sent from Kamailio to Thomson phone does not trigger any response. There are various possible reasons:<br>
<br>
1. INVITE is ignored by Thomson phone<br>
2. INVITE does not make it thorugh to the Thomson phone<br>
2.1 either sent to the wrong port<br>
2.2 or the NAT binding time out, thus NAT does not forward correctly<br>
<br>
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.<br>
<br>
The problem could also be caused by a buggy NAT router or VPN client or firewall ALGs.<br>
<br>
To further debug this issue you could also try the Thomson phone with another VoIP service (e.g. <a href="http://iptel.org" target="_blank">iptel.org</a> or <a href="http://ekiga.net" target="_blank">ekiga.net</a>) or try the Thomson phone from another access (e.g. try it at home bypassing your company FW/NAT).<br>
<br>
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.<div><br>
<br>
regards<br>
klaus<br>
<br>
<br>
Samuel Muller schrieb:<br>
</div><div><div></div><div><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<br>
<br>
On Mon, Dec 1, 2008 at 12:24 PM, Klaus Darilion <<a href="mailto:klaus.mailinglists@pernau.at" target="_blank">klaus.mailinglists@pernau.at</a> <mailto:<a href="mailto:klaus.mailinglists@pernau.at" target="_blank">klaus.mailinglists@pernau.at</a>>> wrote:<br>
<br>
<br>
<br>
Samuel Muller schrieb:<br>
<br>
Hey Klaus,<br>
<br>
first, some answers :<br>
<br>
-> when a thomson is the callee, there's no ringing even if<br>
indicated into the SIP message.<br>
-> when a thomson is the caller, no problem, there's a ring, and<br>
the call is ok with audio.<br>
<br>
<br>
Please be a bit more specific: What does "no ring" mean?<br>
No "180 ringing" response from callee to caller?<br>
"180 ringing" response but no "ring-back" at the caller's client?<br>
<br>
<br>
oups, sorry, I mean : SIP messages are ok, there is all the sig process.<br>
<br>
the architecture is :<br>
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.<br>
<br>
you have the 100 trying, 180 ringing in the SIP message, but there's no ring-back tone for the callee.<br>
<br>
in the attached file :<br>
linksys to linksys, where all the call process is ok (sip + rtp)<br>
thomson to linksys, idem<br>
linksys to thomson, sip ok but rtp apparently not.<br>
I forgot the firewall for the vpn, rtp proxying is required, sorry - so yes rtp proxy must be used.<br>
<br>
regards,<br>
<br>
.Sam.<br>
<br>
</blockquote>
</div></div></blockquote></div>
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