thanks for the reply. if i config the extension in softphone it works fine. but with snom its not working
Bet Regards, Madushan On Fri, Sep 9, 2016 at 10:31 PM, Madushan Geethanga <[email protected] > wrote: > 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? 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