On Thu, Dec 4, 2008 at 1:36 PM, Klaus Darilion <klaus.mailinglists@pernau.at> wrote:

Does the Thoms phone have a logging interface (maybe pcap like SNOM phones, or syslog ...).

Then you could verify if the Thomson phone "sees" the INVITE

klaus

Samuel Muller schrieb:

Hello,

I tried all the ways you told :

I moved the SIP phone at home, which I don't have any firewall and it does not pass through the entreprise fw.
so the SIP phone is directly connected to the proxy.

it registers well, no pbm, but the problem stay.
Impossible to make a call to the Thomson.

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.
All the others SIP stuff are working (Linksys to SJ Phone, ...).

I tried many configuration changes into the Thomson ST2030, unsuccessfully.
I mean it's not a NAT problem ...

Here you are the SIP messages in the kamailio debug from SJ Phone (0123451011) to the f***in' Thomson (0123451014) :

Dec 1 20:49:36 kamailio[29592]: -> incoming SIP buffer message:

INVITE sip:0123451014@sip.720.fr <mailto:sip%3A0123451014@sip.720.fr> SIP/2.0
Via: SIP/2.0/UDP 192.168.1.3 <http://192.168.1.3>;rport;branch=z9hG4bKc0a801030000004549343fcf39c16ac5000000f0
Content-Length: 264
Contact: <sip:0123451011@192.168.1.3:5060 <http://sip:0123451011@192.168.1.3:5060>>
Call-ID: 2E029B5E-1DD2-11B2-A585-993A960A9D75@192.168.1.3 <mailto:2E029B5E-1DD2-11B2-A585-993A960A9D75@192.168.1.3>

Content-Type: application/sdp
CSeq: 2 INVITE
From: "sambook"<sip:0123451011@sip.720.fr <mailto:sip%3A0123451011@sip.720.fr>>;tag=409529589751851917
Max-Forwards: 70
To: <sip:0123451014@sip.720.fr <mailto:sip%3A0123451014@sip.720.fr>>

User-Agent: SJphone/1.60.299a/L (SJ Labs)
Proxy-Authorization: Digest username="0123451011",realm="sip.720.fr <http://sip.720.fr>",
nonce="493440fb0000001024641d0bca47789c4c6f68d81262f201",uri="sip:0123451014@sip.720.fr <mailto:sip%3A0123451014@sip.720.fr>",

response="b8df3912d21ddd8aca40c0bf254bbdcf",cnonce="40952964931016109891",qop="auth",nc="00000001"

v=0
o=- 3437149775 3437149775 IN IP4 192.168.1.3 <http://192.168.1.3>
s=SJphone
c=IN IP4 192.168.1.3 <http://192.168.1.3>

t=0 0
a=direction:active
m=audio 49168 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000

Dec 1 20:49:36 kamailio[29592]: -> outgoing SIP buffer message:

INVITE sip:0123451014@sip.720.fr <mailto:sip%3A0123451014@sip.720.fr> SIP/2.0
Via: SIP/2.0/UDP 192.168.1.3 <http://192.168.1.3>;rport;branch=z9hG4bKc0a801030000004549343fcf39c16ac5000000f0
Content-Length: 264
Contact: <sip:0123451011@192.168.1.3:5060 <http://sip:0123451011@192.168.1.3:5060>>
Call-ID: 2E029B5E-1DD2-11B2-A585-993A960A9D75@192.168.1.3 <mailto:2E029B5E-1DD2-11B2-A585-993A960A9D75@192.168.1.3>

Content-Type: application/sdp
CSeq: 2 INVITE
From: "sambook"<sip:0123451011@sip.720.fr <mailto:sip%3A0123451011@sip.720.fr>>;tag=409529589751851917
Max-Forwards: 69
To: <sip:0123451014@sip.720.fr <mailto:sip%3A0123451014@sip.720.fr>>

User-Agent: SJphone/1.60.299a/L (SJ Labs)
Proxy-Authorization: Digest username="0123451011",realm="sip.720.fr <http://sip.720.fr>",
nonce="493440fb0000001024641d0bca47789c4c6f68d81262f201",uri="sip:0123451014@sip.720.fr <mailto:sip%3A0123451014@sip.720.fr>",

response="b8df3912d21ddd8aca40c0bf254bbdcf",cnonce="40952964931016109891",qop="auth",nc="00000001"

v=0
o=- 3437149775 3437149775 IN IP4 192.168.1.3 <http://192.168.1.3>
s=SJphone
c=IN IP4 192.168.1.3 <http://192.168.1.3>

t=0 0
a=direction:active
m=audio 49168 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000

In the attached file, the full kamailio debug level 9.

It seems that nothing is coming to the Thomson (I don't have any hub where I can sniff the frames).

"The truth is out there" ... :/

.desperate house Sam.

On Mon, Dec 1, 2008 at 1:44 PM, Samuel Muller <sml@720.fr <mailto:sml@720.fr>> wrote:

many thanks Klaus,

I'll check tonight at home, and will reply to you after.

sincerely, thanks !

.Sam.

On Mon, Dec 1, 2008 at 1:28 PM, Klaus Darilion
<klaus.mailinglists@pernau.at <mailto:klaus.mailinglists@pernau.at>>
wrote:

Hi Samuel!

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 firewall ALGs.

To further debug this issue you could also try the Thomson phone
with another VoIP service (e.g. iptel.org <http://iptel.org> or
ekiga.net <http://ekiga.net>) or try the Thomson phone from

another access (e.g. try it at home bypassing your company FW/NAT).

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.

regards
klaus

Samuel Muller schrieb:

On Mon, Dec 1, 2008 at 12:24 PM, Klaus Darilion
<klaus.mailinglists@pernau.at
<mailto:klaus.mailinglists@pernau.at>
<mailto:klaus.mailinglists@pernau.at
<mailto:klaus.mailinglists@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.

regards,

.Sam.