I didnrt say "audio", but all the stuff I saw said "audio".
The siptrace might say "what" exactly, 488 makes me think audio. Snom firmware is "perfect". They only enhance it. It must not have any fixes needed. I stopped trying to get them to pay attention years ago. If they were interested in selling product outside of the ms platform in real numbers, they'd pay attention... ============================ Tony Graziano, Manager Telephone: 434.984.8430 Fax: 434.984.8431 Email: [email protected] LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 Fax: 434.984.8427 Helpdesk Contract Customers: http://www.myitdepartment.net/gethelp/ ----- Original Message ----- From: [email protected] <[email protected]> To: sipXecs developer discussions <[email protected]> Cc: [email protected] <[email protected]> Sent: Mon Oct 04 19:01:14 2010 Subject: Re: [sipx-users] [sipx-dev] Snom 820 anyone? And it does not look like they intend to fix that bug. Michael Jerris schrieb: > On Web page of the snome phone In Identity X/RTP/ RTP/SAVP: Set to > Optional > > funny story about this one, the guys at snom filed a bug in freeswitch > a while back complaining that we accepted a=crypto in an m=audio > RTP/AVP section, saying we should always 488 that call as a=crypto > should only be in SAVP. Turns out their default config is to send it > wrong, and the only device we have issues with on this is the Snom phones. > > Mike > > On Oct 4, 2010, at 5:09 PM, George Niculae wrote: > >> I have a Snom820 phone registered but I'm getting a 488 Not Acceptable >> Here while trying to join conference (from X-Lite works just fine). I >> suspect something wrong with firmware but not sure... snom820-SIP >> 8.1.2 15856. Please find below INVITE message and 488 one (removed >> relevant ips). >> >> Thanks in advance for any pointer, >> George >> >> Sent to tcp:x.x.x.x:5060 at 4/10/2010 22:56:12:119 (1423 bytes): >> >> INVITE sip:[email protected];user=phone SIP/2.0 >> Via: SIP/2.0/TCP 192.168.0.1:1029;branch=z9hG4bK-4oh6gt1acd3g;rport >> From: "George" <sip:[email protected]>;tag=1nozggwi04 >> To: <sip:[email protected];user=phone> >> Call-ID: 3c2670b590cb-lcifsq0o4kpa >> CSeq: 2 INVITE >> Max-Forwards: 70 >> Contact: <"sip:[email protected];gr >> <sip:%7e%7egr%[email protected];gr>">;reg-id=1 >> P-Key-Flags: resolution="31x13", keys="4" >> User-Agent: snom820/8.1.2 >> Accept: application/sdp >> Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, >> PRACK, MESSAGE, INFO >> Allow-Events: talk, hold, refer, call-info >> Supported: timer, 100rel, replaces, from-change >> Session-Expires: 3600;refresher=uas >> Min-SE: 90 >> Proxy-Authorization: Digest >> username="201",realm="test.com >> <http://test.com>",nonce="41809edfe611c2e0a63879e4523183624caa315b",uri="sip:[email protected];user=phone",qop=auth,nc=00000001,cnonce="046a7f70",response="22d8b718a3a5823f06542c75d9478edf",algorithm=MD5 >> Content-Type: application/sdp >> Content-Length: 456 >> >> v=0 >> o=root 1750241528 1750241528 IN IP4 192.168.0.1 >> s=call >> c=IN IP4 192.168.0.1 >> t=0 0 >> m=audio 52318 RTP/AVP 0 8 9 99 3 18 4 101 >> a=crypto:1 AES_CM_128_HMAC_SHA1_32 >> inline:uvUGQchyOdivmjoHUvUlFOSg+rmBnCBJWrAyw9wT >> a=rtpmap:0 pcmu/8000 >> a=rtpmap:8 pcma/8000 >> a=rtpmap:9 g722/8000 >> a=rtpmap:99 g726-32/8000 >> a=rtpmap:3 gsm/8000 >> a=rtpmap:18 g729/8000 >> a=rtpmap:4 g723/8000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-16 >> a=ptime:20 >> a=sendrecv >> >> ------------------------------- >> >> Received from tcp:x.x.x.x:5060 at 4/10/2010 22:56:13:147 (752 bytes): >> >> SIP/2.0 488 Not Acceptable Here >> Via: SIP/2.0/TCP >> 192.168.0.1:1029;branch=z9hG4bK-4oh6gt1acd3g;rport=64525;received=x.x.x.x >> From: "George" <sip:[email protected]>;tag=1nozggwi04 >> To: <sip:[email protected];user=phone>;tag=y9D60v174Q7gm >> Call-Id: 3c2670b590cb-lcifsq0o4kpa >> Cseq: 2 INVITE >> User-Agent: FreeSWITCH-mod_sofia/1.0.head-17188.16739.2 >> Accept: application/sdp >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >> REGISTER, REFER, NOTIFY >> Supported: timer, precondition, path, replaces >> Allow-Events: talk, refer >> Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION" >> Content-Length: 0 >> Remote-Party-Id: "Conf" >> <sip:[email protected]>;party=calling;privacy=off;screen=no >> Date: Mon, 04 Oct 2010 19:56:12 GMT >> _______________________________________________ >> sipx-dev mailing list >> [email protected] <mailto:[email protected]> >> List Archive: http://list.sipfoundry.org/archive/sipx-dev/ > > ------------------------------------------------------------------------ > > _______________________________________________ > sipx-dev mailing list > [email protected] > List Archive: http://list.sipfoundry.org/archive/sipx-dev/ _______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users/ _______________________________________________ sipx-dev mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-dev/
