You didn't say which version of Asterisk you were using. insecure=very is deprecated in favor of insecure=port,invite
Many of the VoIP providers do not have this right in their examples. Darrick On 11/08/2010 05:52 PM, Gregory Malsack wrote: > Not sure if you read the configs I attached, but that line is already in > there... Guess again... > > > -----Original Message----- > From: [email protected] > [mailto:[email protected]] 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<[email protected]> 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:[email protected]:5060 SIP/2.0 >> >> Call-ID: [email protected] >> >> CSeq: 1 INVITE >> >> From: "Wi M"<sip:[email protected];user=phone>;tag=9bbc >> >> To: "Gregory Malsack"<sip:[email protected]> >> >> 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:[email protected]: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 - [email protected] >> >> [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:[email protected];user=phone>;tag=9bbc >> >> To: "Gregory Malsack"<sip:[email protected]>;tag=as4fffe111 >> >> Call-ID: [email protected] >> >> 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 '[email protected]' 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:[email protected]:5060 SIP/2.0 >> >> Call-ID: [email protected] >> >> CSeq: 1 ACK >> >> From: "Wi M"<sip:number [email protected];user=phone>;tag=9bbc >> >> To: "username"<sip:[email protected]>;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 >> > -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? 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