[Kamailio-Users] Thomson ST2030 SIP contact problem
Klaus Darilion
klaus.mailinglists at pernau.at
Thu Dec 4 13:36:57 CET 2008
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 at sip.720.fr <mailto:sip%3A0123451014 at 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 at 192.168.1.3:5060
> <http://sip:0123451011@192.168.1.3:5060>>
> Call-ID: 2E029B5E-1DD2-11B2-A585-993A960A9D75 at 192.168.1.3
> <mailto:2E029B5E-1DD2-11B2-A585-993A960A9D75 at 192.168.1.3>
> Content-Type: application/sdp
> CSeq: 2 INVITE
> From: "sambook"<sip:0123451011 at sip.720.fr
> <mailto:sip%3A0123451011 at sip.720.fr>>;tag=409529589751851917
> Max-Forwards: 70
> To: <sip:0123451014 at sip.720.fr <mailto:sip%3A0123451014 at 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 at sip.720.fr
> <mailto:sip%3A0123451014 at 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 at sip.720.fr <mailto:sip%3A0123451014 at 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 at 192.168.1.3:5060
> <http://sip:0123451011@192.168.1.3:5060>>
> Call-ID: 2E029B5E-1DD2-11B2-A585-993A960A9D75 at 192.168.1.3
> <mailto:2E029B5E-1DD2-11B2-A585-993A960A9D75 at 192.168.1.3>
> Content-Type: application/sdp
> CSeq: 2 INVITE
> From: "sambook"<sip:0123451011 at sip.720.fr
> <mailto:sip%3A0123451011 at sip.720.fr>>;tag=409529589751851917
> Max-Forwards: 69
> To: <sip:0123451014 at sip.720.fr <mailto:sip%3A0123451014 at 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 at sip.720.fr
> <mailto:sip%3A0123451014 at 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 at 720.fr
> <mailto:sml at 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 at pernau.at <mailto:klaus.mailinglists at 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 at pernau.at
> <mailto:klaus.mailinglists at pernau.at>
> <mailto: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.
>
> regards,
>
> .Sam.
>
>
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