Yep, I can use the CLI functionnalities.<br>I reply asap :)<br><br>thanks Klaus,<br><br>.Sam.<br><br><br><div class="gmail_quote">On Thu, Dec 4, 2008 at 1:36 PM, Klaus Darilion <span dir="ltr"><<a href="mailto:klaus.mailinglists@pernau.at">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;">Does the Thoms phone have a logging interface (maybe pcap like SNOM phones, or syslog ...).<br>
<br>
Then you could verify if the Thomson phone "sees" the INVITE<br>
<br>
klaus<br>
<br>
Samuel Muller schrieb:<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><div class="Ih2E3d">
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></div>
INVITE <a href="mailto:sip%3A0123451014@sip.720.fr" target="_blank">sip:0123451014@sip.720.fr</a> <mailto:<a href="mailto:sip%253A0123451014@sip.720.fr" target="_blank">sip%3A0123451014@sip.720.fr</a>> SIP/2.0<br>
Via: SIP/2.0/UDP <a href="http://192.168.1.3" target="_blank">192.168.1.3</a> <<a href="http://192.168.1.3" target="_blank">http://192.168.1.3</a>>;rport;branch=z9hG4bKc0a801030000004549343fcf39c16ac5000000f0<br>
Content-Length: 264<br>
Contact: <<a href="http://sip:0123451011@192.168.1.3:5060" target="_blank">sip:0123451011@192.168.1.3:5060</a> <<a href="http://sip:0123451011@192.168.1.3:5060" target="_blank">http://sip:0123451011@192.168.1.3:5060</a>>><br>
Call-ID: <a href="mailto:2E029B5E-1DD2-11B2-A585-993A960A9D75@192.168.1.3" target="_blank">2E029B5E-1DD2-11B2-A585-993A960A9D75@192.168.1.3</a> <mailto:<a href="mailto:2E029B5E-1DD2-11B2-A585-993A960A9D75@192.168.1.3" target="_blank">2E029B5E-1DD2-11B2-A585-993A960A9D75@192.168.1.3</a>><div class="Ih2E3d">
<br>
Content-Type: application/sdp<br>
CSeq: 2 INVITE<br></div>
From: "sambook"<<a href="mailto:sip%3A0123451011@sip.720.fr" target="_blank">sip:0123451011@sip.720.fr</a> <mailto:<a href="mailto:sip%253A0123451011@sip.720.fr" target="_blank">sip%3A0123451011@sip.720.fr</a>>>;tag=409529589751851917<br>
Max-Forwards: 70<br>
To: <<a href="mailto:sip%3A0123451014@sip.720.fr" target="_blank">sip:0123451014@sip.720.fr</a> <mailto:<a href="mailto:sip%253A0123451014@sip.720.fr" target="_blank">sip%3A0123451014@sip.720.fr</a>>><div class="Ih2E3d">
<br>
User-Agent: SJphone/1.60.299a/L (SJ Labs)<br></div>
Proxy-Authorization: Digest username="0123451011",realm="<a href="http://sip.720.fr" target="_blank">sip.720.fr</a> <<a href="http://sip.720.fr" target="_blank">http://sip.720.fr</a>>",<br>
nonce="493440fb0000001024641d0bca47789c4c6f68d81262f201",uri="<a href="mailto:sip%3A0123451014@sip.720.fr" target="_blank">sip:0123451014@sip.720.fr</a> <mailto:<a href="mailto:sip%253A0123451014@sip.720.fr" target="_blank">sip%3A0123451014@sip.720.fr</a>>",<div class="Ih2E3d">
<br>
response="b8df3912d21ddd8aca40c0bf254bbdcf",cnonce="40952964931016109891",qop="auth",nc="00000001"<br>
<br>
v=0<br></div>
o=- 3437149775 3437149775 IN IP4 <a href="http://192.168.1.3" target="_blank">192.168.1.3</a> <<a href="http://192.168.1.3" target="_blank">http://192.168.1.3</a>><br>
s=SJphone<br>
c=IN IP4 <a href="http://192.168.1.3" target="_blank">192.168.1.3</a> <<a href="http://192.168.1.3" target="_blank">http://192.168.1.3</a>><div class="Ih2E3d"><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></div>
INVITE <a href="mailto:sip%3A0123451014@sip.720.fr" target="_blank">sip:0123451014@sip.720.fr</a> <mailto:<a href="mailto:sip%253A0123451014@sip.720.fr" target="_blank">sip%3A0123451014@sip.720.fr</a>> SIP/2.0<br>
Via: SIP/2.0/UDP <a href="http://192.168.1.3" target="_blank">192.168.1.3</a> <<a href="http://192.168.1.3" target="_blank">http://192.168.1.3</a>>;rport;branch=z9hG4bKc0a801030000004549343fcf39c16ac5000000f0<br>
Content-Length: 264<br>
Contact: <<a href="http://sip:0123451011@192.168.1.3:5060" target="_blank">sip:0123451011@192.168.1.3:5060</a> <<a href="http://sip:0123451011@192.168.1.3:5060" target="_blank">http://sip:0123451011@192.168.1.3:5060</a>>><br>
Call-ID: <a href="mailto:2E029B5E-1DD2-11B2-A585-993A960A9D75@192.168.1.3" target="_blank">2E029B5E-1DD2-11B2-A585-993A960A9D75@192.168.1.3</a> <mailto:<a href="mailto:2E029B5E-1DD2-11B2-A585-993A960A9D75@192.168.1.3" target="_blank">2E029B5E-1DD2-11B2-A585-993A960A9D75@192.168.1.3</a>><div class="Ih2E3d">
<br>
Content-Type: application/sdp<br>
CSeq: 2 INVITE<br></div>
From: "sambook"<<a href="mailto:sip%3A0123451011@sip.720.fr" target="_blank">sip:0123451011@sip.720.fr</a> <mailto:<a href="mailto:sip%253A0123451011@sip.720.fr" target="_blank">sip%3A0123451011@sip.720.fr</a>>>;tag=409529589751851917<br>
Max-Forwards: 69<br>
To: <<a href="mailto:sip%3A0123451014@sip.720.fr" target="_blank">sip:0123451014@sip.720.fr</a> <mailto:<a href="mailto:sip%253A0123451014@sip.720.fr" target="_blank">sip%3A0123451014@sip.720.fr</a>>><div class="Ih2E3d">
<br>
User-Agent: SJphone/1.60.299a/L (SJ Labs)<br></div>
Proxy-Authorization: Digest username="0123451011",realm="<a href="http://sip.720.fr" target="_blank">sip.720.fr</a> <<a href="http://sip.720.fr" target="_blank">http://sip.720.fr</a>>",<br>
nonce="493440fb0000001024641d0bca47789c4c6f68d81262f201",uri="<a href="mailto:sip%3A0123451014@sip.720.fr" target="_blank">sip:0123451014@sip.720.fr</a> <mailto:<a href="mailto:sip%253A0123451014@sip.720.fr" target="_blank">sip%3A0123451014@sip.720.fr</a>>",<div class="Ih2E3d">
<br>
response="b8df3912d21ddd8aca40c0bf254bbdcf",cnonce="40952964931016109891",qop="auth",nc="00000001"<br>
<br>
v=0<br></div>
o=- 3437149775 3437149775 IN IP4 <a href="http://192.168.1.3" target="_blank">192.168.1.3</a> <<a href="http://192.168.1.3" target="_blank">http://192.168.1.3</a>><br>
s=SJphone<br>
c=IN IP4 <a href="http://192.168.1.3" target="_blank">192.168.1.3</a> <<a href="http://192.168.1.3" target="_blank">http://192.168.1.3</a>><div class="Ih2E3d"><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><div class="Ih2E3d">
On Mon, Dec 1, 2008 at 1:44 PM, Samuel Muller <<a href="mailto:sml@720.fr" target="_blank">sml@720.fr</a> <mailto:<a href="mailto:sml@720.fr" target="_blank">sml@720.fr</a>>> wrote:<br>
<br>
many thanks Klaus,<br>
<br>
I'll check tonight at home, and will reply to you after.<br>
<br>
sincerely, thanks !<br>
<br>
.Sam.<br>
<br>
<br>
<br>
<br>
On Mon, Dec 1, 2008 at 1:28 PM, Klaus Darilion<br></div><div class="Ih2E3d">
<<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>>><br>
wrote:<br>
<br></div><div class="Ih2E3d">
Hi Samuel!<br>
<br>
The INVITE sent from Kamailio to Thomson phone does not trigger<br>
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<br>
correctly<br>
<br>
Thus, verify if the INVITE is received by the NAT device and<br>
forwarded to the Thomson phone (e.g. putting a hub between the<br>
NAT router and the phone). REGISTER with the Thomson phone and<br>
then immediately after call it (linksys->thomson) - this should<br>
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<br>
client or firewall ALGs.<br>
<br>
To further debug this issue you could also try the Thomson phone<br></div>
with another VoIP service (e.g. <a href="http://iptel.org" target="_blank">iptel.org</a> <<a href="http://iptel.org" target="_blank">http://iptel.org</a>> or<br>
<a href="http://ekiga.net" target="_blank">ekiga.net</a> <<a href="http://ekiga.net" target="_blank">http://ekiga.net</a>>) or try the Thomson phone from<div><div></div><div class="Wj3C7c"><br>
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<br>
port 5678 and also use other ports locally for the SIP clients.<br>
SIP ALGs (application level gateways) usually are triggered by<br>
port 5060.<br>
<br>
<br>
regards<br>
klaus<br>
<br>
<br>
Samuel Muller schrieb:<br>
<br>
<br>
<br>
On Mon, Dec 1, 2008 at 12:24 PM, Klaus Darilion<br>
<<a href="mailto:klaus.mailinglists@pernau.at" target="_blank">klaus.mailinglists@pernau.at</a><br>
<mailto:<a href="mailto:klaus.mailinglists@pernau.at" target="_blank">klaus.mailinglists@pernau.at</a>><br>
<mailto:<a href="mailto:klaus.mailinglists@pernau.at" target="_blank">klaus.mailinglists@pernau.at</a><br>
<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<br>
even if<br>
indicated into the SIP message.<br>
-> when a thomson is the caller, no problem, there's<br>
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<br>
caller's client?<br>
<br>
<br>
oups, sorry, I mean : SIP messages are ok, there is all the<br>
sig process.<br>
<br>
the architecture is :<br>
linksys + thomson -> cisco 827 -> SDSL -> our backbone which<br>
have a firewall for VPN (so NAT and NAPT are applied here),<br>
then the kamailio with a public ip.<br>
<br>
you have the 100 trying, 180 ringing in the SIP message, but<br>
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<br>
required, sorry - so yes rtp proxy must be used.<br>
<br>
regards,<br>
<br>
.Sam.<br>
<br>
<br>
</div></div></blockquote>
</blockquote></div>