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
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