Administrator TOOTAI
2016-Sep-09 16:57 UTC
[asterisk-users] Asterisk 13 PJSIP with Snom 710
Le 09/09/2016 ? 18:32, Madushan Geethanga a ?crit :> 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:0 at 54.206.59.252 <mailto:sip%3A0 at 54.206.59.252>;user=phone SIP/2.0 > Via: SIP/2.0/UDP 123.231.72.210:45835;branch=z9hG4bK-bskkkx1t5bas;rport > From: "outburns00-nhvg5vjjn6-2001" > <sip:outburns00-nhvg5vjjn6-2001 at 54.206.59.252 > <mailto:sip%3Aoutburns00-nhvg5vjjn6-2001 at 54.206.59.252>>;tag=1bb809zgaa > To: <sip:0 at 54.206.59.252 <mailto:sip%3A0 at 54.206.59.252>;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:outburns00-nhvg5vjjn6-2001 at 123.231.72.210:45835 > <http://sip:outburns00-nhvg5vjjn6-2001 at 123.231.72.210:45835>>;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;branch=z9hG4bK-bskkkx1t5bas > Call-ID: 313437333433383639323238313539-ahn3begiq66q > From: "outburns00-nhvg5vjjn6-2001" > <sip:outburns00-nhvg5vjjn6-2001 at 54.206.59.252 > <mailto:sip%3Aoutburns00-nhvg5vjjn6-2001 at 54.206.59.252>>;tag=1bb809zgaa > To: <sip:0 at 54.206.59.252 > <mailto:sip%3A0 at 54.206.59.252>;user=phone>;tag=z9hG4bK-bskkkx1t5bas > CSeq: 1 INVITE > WWW-Authenticate: Digest > realm="asterisk",nonce="1473438693/ef923d25464dbedc1dbd85e0ccea08b7",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:0 at 54.206.59.252 <mailto:sip%3A0 at 54.206.59.252>;user=phone SIP/2.0 > Via: SIP/2.0/UDP 123.231.72.210:45835;branch=z9hG4bK-bskkkx1t5bas;rport > From: "outburns00-nhvg5vjjn6-2001" > <sip:outburns00-nhvg5vjjn6-2001 at 54.206.59.252 > <mailto:sip%3Aoutburns00-nhvg5vjjn6-2001 at 54.206.59.252>>;tag=1bb809zgaa > To: <sip:0 at 54.206.59.252 > <mailto:sip%3A0 at 54.206.59.252>;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:outburns00-nhvg5vjjn6-2001 at 123.231.72.210:45835 > <http://sip:outburns00-nhvg5vjjn6-2001 at 123.231.72.210:45835>>;reg-id=1 > Content-Length: 0 > > > Best Regards, > Madushan > > > > On Fri, Sep 9, 2016 at 9:53 PM, Madushan Geethanga > <mgliyanage.rc at gmail.com <mailto:mgliyanage.rc at gmail.com>> 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 > > > >
yes I have unchecked it. On Fri, Sep 9, 2016 at 10:27 PM, Administrator TOOTAI <admin at tootai.net> wrote:> Le 09/09/2016 ? 18:32, Madushan Geethanga a ?crit : > >> 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:0 at 54.206.59.252 <mailto:sip%3A0 at 54.206.59.252>;user=phone >> SIP/2.0 >> Via: SIP/2.0/UDP 123.231.72.210:45835;branch=z9hG4bK-bskkkx1t5bas;rport >> From: "outburns00-nhvg5vjjn6-2001" >> <sip:outburns00-nhvg5vjjn6-2001 at 54.206.59.252 >> <mailto:sip%3Aoutburns00-nhvg5vjjn6-2001 at 54.206.59.252>>;tag=1bb809zgaa >> To: <sip:0 at 54.206.59.252 <mailto:sip%3A0 at 54.206.59.252>;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:outburns00-nhvg5vjjn6-2001 at 123.231.72.210:45835 >> <http://sip:outburns00-nhvg5vjjn6-2001 at 123.231.72.210:45835>>;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:outburns00-nhvg5vjjn6-2001 at 54.206.59.252 >> <mailto:sip%3Aoutburns00-nhvg5vjjn6-2001 at 54.206.59.252>>;tag=1bb809zgaa >> To: <sip:0 at 54.206.59.252 >> <mailto:sip%3A0 at 54.206.59.252>;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:0 at 54.206.59.252 <mailto:sip%3A0 at 54.206.59.252>;user=phone SIP/2.0 >> Via: SIP/2.0/UDP 123.231.72.210:45835;branch=z9hG4bK-bskkkx1t5bas;rport >> From: "outburns00-nhvg5vjjn6-2001" >> <sip:outburns00-nhvg5vjjn6-2001 at 54.206.59.252 >> <mailto:sip%3Aoutburns00-nhvg5vjjn6-2001 at 54.206.59.252>>;tag=1bb809zgaa >> To: <sip:0 at 54.206.59.252 >> <mailto:sip%3A0 at 54.206.59.252>;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:outburns00-nhvg5vjjn6-2001 at 123.231.72.210:45835 >> <http://sip:outburns00-nhvg5vjjn6-2001 at 123.231.72.210:45835>>;reg-id=1 >> Content-Length: 0 >> >> >> Best Regards, >> Madushan >> >> >> >> On Fri, Sep 9, 2016 at 9:53 PM, Madushan Geethanga >> <mgliyanage.rc at gmail.com <mailto:mgliyanage.rc at gmail.com>> 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 >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160909/74f259ce/attachment.html>
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 <mgliyanage.rc at gmail.com> wrote:> yes I have unchecked it. > > On Fri, Sep 9, 2016 at 10:27 PM, Administrator TOOTAI <admin at tootai.net> > wrote: > >> Le 09/09/2016 ? 18:32, Madushan Geethanga a ?crit : >> >>> 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:0 at 54.206.59.252 <mailto:sip%3A0 at 54.206.59.252>;user=phone >>> SIP/2.0 >>> Via: SIP/2.0/UDP 123.231.72.210:45835;branch=z9hG4bK-bskkkx1t5bas;rport >>> From: "outburns00-nhvg5vjjn6-2001" >>> <sip:outburns00-nhvg5vjjn6-2001 at 54.206.59.252 >>> <mailto:sip%3Aoutburns00-nhvg5vjjn6-2001 at 54.206.59.252>>;tag=1bb809zgaa >>> To: <sip:0 at 54.206.59.252 <mailto:sip%3A0 at 54.206.59.252>;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:outburns00-nhvg5vjjn6-2001 at 123.231.72.210:45835 >>> <http://sip:outburns00-nhvg5vjjn6-2001 at 123.231.72.210:45835>>;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:outburns00-nhvg5vjjn6-2001 at 54.206.59.252 >>> <mailto:sip%3Aoutburns00-nhvg5vjjn6-2001 at 54.206.59.252>>;tag=1bb809zgaa >>> To: <sip:0 at 54.206.59.252 >>> <mailto:sip%3A0 at 54.206.59.252>;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:0 at 54.206.59.252 <mailto:sip%3A0 at 54.206.59.252>;user=phone >>> SIP/2.0 >>> Via: SIP/2.0/UDP 123.231.72.210:45835;branch=z9hG4bK-bskkkx1t5bas;rport >>> From: "outburns00-nhvg5vjjn6-2001" >>> <sip:outburns00-nhvg5vjjn6-2001 at 54.206.59.252 >>> <mailto:sip%3Aoutburns00-nhvg5vjjn6-2001 at 54.206.59.252>>;tag=1bb809zgaa >>> To: <sip:0 at 54.206.59.252 >>> <mailto:sip%3A0 at 54.206.59.252>;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:outburns00-nhvg5vjjn6-2001 at 123.231.72.210:45835 >>> <http://sip:outburns00-nhvg5vjjn6-2001 at 123.231.72.210:45835>>;reg-id=1 >>> Content-Length: 0 >>> >>> >>> Best Regards, >>> Madushan >>> >>> >>> >>> On Fri, Sep 9, 2016 at 9:53 PM, Madushan Geethanga >>> <mgliyanage.rc at gmail.com <mailto:mgliyanage.rc at gmail.com>> 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 >> > >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160909/a762a797/attachment.html>