yes I have unchecked it. On Fri, Sep 9, 2016 at 10:27 PM, Administrator TOOTAI <[email protected]> wrote:
> Le 09/09/2016 à 18:32, Madushan Geethanga a écrit : > >> Hi, >> > > If you're not using RTP encryption did you uncheck the option in your RTP > TAB from identity ? > > >> This is the log. ex dialling 0 from snom phone >> >> >> <--- Received SIP request (1230 bytes) from UDP:123.231.72.210:33878 >> <http://123.231.72.210:33878> ---> >> INVITE sip:[email protected] <mailto:sip%[email protected]>;user=phone >> SIP/2.0 >> Via: SIP/2.0/UDP 123.231.72.210:45835;branch=z9hG4bK-bskkkx1t5bas;rport >> From: "outburns00-nhvg5vjjn6-2001" >> <sip:[email protected] >> <mailto:sip%[email protected]>>;tag=1bb809zgaa >> To: <sip:[email protected] <mailto:sip%[email protected]>;user=phone> >> Call-ID: 313437333433383639323238313539-ahn3begiq66q >> CSeq: 1 INVITE >> Max-Forwards: 70 >> User-Agent: snom710/8.7.5.35 <http://8.7.5.35> >> Contact: <sip:[email protected]:45835 >> <http://sip:[email protected]:45835>>;reg-id=1 >> >> X-Serialnumber: 000413747C96 >> P-Key-Flags: resolution="31x13", keys="4" >> Accept: application/sdp >> Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, >> PRACK, MESSAGE, INFO, UPDATE >> Allow-Events: talk, hold, refer, call-info >> Supported: timer, 100rel, replaces, from-change >> Session-Expires: 3600 >> Min-SE: 90 >> Content-Type: application/sdp >> Content-Length: 405 >> >> v=0 >> o=root 2136927789 2136927789 IN IP4 192.168.2.28 >> s=call >> c=IN IP4 123.231.72.210 >> t=0 0 >> m=audio 62724 RTP/AVP 9 0 8 3 99 112 18 101 >> a=rtpmap:9 G722/8000 >> a=rtpmap:0 PCMU/8000 >> a=rtpmap:8 PCMA/8000 >> a=rtpmap:3 GSM/8000 >> a=rtpmap:99 G726-32/8000 >> a=rtpmap:112 AAL2-G726-32/8000 >> a=rtpmap:18 G729/8000 >> a=fmtp:18 annexb=no >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-15 >> a=ptime:20 >> a=sendrecv >> >> <--- Transmitting SIP response (572 bytes) to UDP:123.231.72.210:33878 >> <http://123.231.72.210:33878> ---> >> SIP/2.0 401 Unauthorized >> Via: SIP/2.0/UDP >> 123.231.72.210:45835;rport=33878;received=123.231.72.210;bra >> nch=z9hG4bK-bskkkx1t5bas >> Call-ID: 313437333433383639323238313539-ahn3begiq66q >> From: "outburns00-nhvg5vjjn6-2001" >> <sip:[email protected] >> <mailto:sip%[email protected]>>;tag=1bb809zgaa >> To: <sip:[email protected] >> <mailto:sip%[email protected]>;user=phone>;tag=z9hG4bK-bskkkx1t5bas >> CSeq: 1 INVITE >> WWW-Authenticate: Digest >> realm="asterisk",nonce="1473438693/ef923d25464dbedc1dbd85e0c >> cea08b7",opaque="210b270d7abb2354",algorithm=md5,qop="auth" >> Server: Asterisk PBX certified/13.8-cert2 >> Content-Length: 0 >> >> >> <--- Received SIP request (487 bytes) from UDP:123.231.72.210:33878 >> <http://123.231.72.210:33878> ---> >> ACK sip:[email protected] <mailto:sip%[email protected]>;user=phone SIP/2.0 >> Via: SIP/2.0/UDP 123.231.72.210:45835;branch=z9hG4bK-bskkkx1t5bas;rport >> From: "outburns00-nhvg5vjjn6-2001" >> <sip:[email protected] >> <mailto:sip%[email protected]>>;tag=1bb809zgaa >> To: <sip:[email protected] >> <mailto:sip%[email protected]>;user=phone>;tag=z9hG4bK-bskkkx1t5bas >> Call-ID: 313437333433383639323238313539-ahn3begiq66q >> CSeq: 1 ACK >> Max-Forwards: 70 >> User-Agent: snom710/8.7.5.35 <http://8.7.5.35> >> Contact: <sip:[email protected]:45835 >> <http://sip:[email protected]:45835>>;reg-id=1 >> Content-Length: 0 >> >> >> Best Regards, >> Madushan >> >> >> >> On Fri, Sep 9, 2016 at 9:53 PM, Madushan Geethanga >> <[email protected] <mailto:[email protected]>> wrote: >> >> Hi, >> >> I'm trying to setup snom 710 phone with asterisk 13 with PJSIP. >> inbound is working fine but i cannot dial out. i don't hear anything >> on the phone and asterisk CLI also does not show anything. my config >> is. please advice. >> >> [2001] >> type=endpoint >> context=out-local >> disallow=all >> allow=ulaw >> allow=alaw >> transport=system-udp >> auth=2001 >> aors=2001 >> direct_media=no >> rtp_symmetric=yes >> force_rport=yes >> allow=alaw >> allow=speex >> allow=speex16 >> allow=speex32 >> allow=gsm >> >> >> [2001] >> type=aor >> qualify_frequency=5000 >> authenticate_qualify=yes >> max_contacts=1 >> remove_existing=yes >> >> [2001] >> type=auth >> auth_type=userpass >> password=test >> username=test >> >> Best Regards, >> Madushan >> >> >> >> >> > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016 > http://www.asterisk.org/community/astricon-user-conference > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016 http://www.asterisk.org/community/astricon-user-conference New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
