Not sure if you read the configs I attached, but that line is already in there... Guess again...
-----Original Message----- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of C F Sent: Wednesday, November 03, 2010 7:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] inbound call issue... insecure=very should fix it. On Wed, Nov 3, 2010 at 4:08 AM, Gregory Malsack <gmals...@gmellc.com> wrote: > Can anyone tell me why my inbound calls keep getting rejected with 401? > > > > Here’s the debug information: > > > > > > <--- SIP read from UDP:147.135.32.221:5060 ---> > > INVITE sip:6087294...@216.26.109.22:5060 SIP/2.0 > > Call-ID: 31007e...@147.135.32.221 > > CSeq: 1 INVITE > > From: "Wi M"<sip:4144038...@147.135.32.221;user=phone>;tag=9bbc > > To: "Gregory Malsack"<sip:s...@216.26.109.22> > > Via: SIP/2.0/UDP > 147.135.32.221:5060;branch=z9hG4bK-BroadWorks.192.168.0.3-192.168.32.221V5060-0-138966241-538634340-1288768121281- > > Contact: <sip:4144038...@147.135.32.221:5060> > > Supported: 100rel > > Max-Forwards: 69 > > Content-Length: 308 > > Content-Type: application/sdp > > > > v=0 > > o=2475098871 10 10 IN IP4 147.135.2.247 > > s=- > > c=IN IP4 147.135.2.248 > > t=0 0 > > m=audio 15502 RTP/AVP 0 18 8 96 9 101 > > a=rtpmap:0 PCMU/8000 > > a=rtpmap:18 G729/8000 > > a=fmtp:18 annexb=no > > a=rtpmap:8 PCMA/8000 > > a=rtpmap:96 iLBC/8000 > > a=fmtp:96 mode=30 > > a=rtpmap:9 G722/8000 > > a=rtpmap:101 telephone-event/8000 > > > > <-------------> > > [Nov 3 02:08:40] VERBOSE[7207] chan_sip.c: --- (11 headers 14 lines) --- > > [Nov 3 02:08:40] VERBOSE[7207] netsock.c: == Using SIP RTP CoS mark 5 > > [Nov 3 02:08:40] VERBOSE[7207] chan_sip.c: Sending to 147.135.32.221 : 5060 > (NAT) > > [Nov 3 02:08:40] VERBOSE[7207] chan_sip.c: Using INVITE request as basis > request - 31007e...@147.135.32.221 > > [Nov 3 02:08:40] VERBOSE[7207] chan_sip.c: Found peer 'trunk_1' for > '4144038968' from 147.135.32.221:5060 > > [Nov 3 02:08:40] VERBOSE[7207] chan_sip.c: > > <--- Reliably Transmitting (NAT) to 147.135.32.221:5060 ---> > > SIP/2.0 401 Unauthorized > > Via: SIP/2.0/UDP > 147.135.32.221:5060;branch=z9hG4bK-BroadWorks.192.168.0.3-192.168.32.221V5060-0-138966241-538634340-1288768121281-;received=147.135.32.221 > > From: "Wi M"<sip:4144038...@147.135.32.221;user=phone>;tag=9bbc > > To: "Gregory Malsack"<sip:s...@216.26.109.22>;tag=as4fffe111 > > Call-ID: 31007e...@147.135.32.221 > > CSeq: 1 INVITE > > Server: Asterisk PBX > > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO > > Supported: replaces, timer > > WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2dd58be8" > > Content-Length: 0 > > > > <------------> > > [Nov 3 02:08:40] VERBOSE[7207] chan_sip.c: Scheduling destruction of SIP > dialog '31007e...@147.135.32.221' in 32000 ms (Method: INVITE) > > [Nov 3 02:08:40] VERBOSE[7207] chan_sip.c: > > <--- SIP read from UDP:147.135.32.221:5060 ---> > > ACK sip:6087294...@216.26.109.22:5060 SIP/2.0 > > Call-ID: 31007e...@147.135.32.221 > > CSeq: 1 ACK > > From: "Wi M"<sip:number f...@147.135.32.221;user=phone>;tag=9bbc > > To: "username"<sip:s...@216.26.109.22>;tag=as4fffe111 > > Via: SIP/2.0/UDP > 147.135.32.221:5060;branch=z9hG4bK-BroadWorks.192.168.0.3-192.168.32.221V5060-0-138966241-538634340-1288768121281- > > Max-Forwards: 70 > > Content-Length: 0 > > > > > > > > > > > > Here’s the configs: > > > > subscribecontext = device-hints > > allowexternaldomains = yes > > allowguest = yes > > allowsubscribe = yes > > allowtransfer = yes > > alwaysauthreject = no > > autodomain = no > > callevents = no > > canreinvite = yes > > checkmwi = 10 > > compactheaders = no > > defaultexpiry = 120 > > dumphistory = no > > externip = 216.26.109.22 > > g726nonstandard = no > > jbenable = yes > > jbforce = no > > jblog = no > > localnet = internal subnet > > maxcallbitrate = 384 > > maxexpiry = 3600 > > minexpiry = 60 > > mohinterpret = default > > nat = yes > > notifyringing = yes > > pedantic = no > > progressinband = never > > promiscredir = no > > realm = asterisk > > recordhistory = no > > registerattempts = 0 > > registertimeout = 20 > > relaxdtmf = no > > sendrpid = no > > sipdebug = no > > t1min = 100 > > t38pt_udptl = no > > tos_audio = none > > tos_sip = none > > tos_video = none > > trustrpid = no > > useragent = Asterisk PBX > > usereqphone = no > > videosupport = no > > disallow = all > > allow = ulaw,gsm > > subscribecontext = device-hints > > > > register => 6087294351:<sip password>@sip.broadvoice.com > > > > [trunk_1] > > type=peer > > user=phone > > host=sip.broadvoice.com > > fromdomain=sip.broadvoice.com > > fromuser=6087294351 > > secret=<sip password> > > username=6087294351 > > insecure=very > > context=DID_trunk_1 > > authname=6087294351 > > dtmfmode=inband > > dtmf=inband > > canreinvite=no > > > > [guest] > > type=friend > > host=dynamic > > canreinvite=no > > context=DID_trunk_1 > > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
