[asterisk-users] inbound call issue...

[C F]

insecure=very should fix it.

On Wed, Nov 3, 2010 at 4:08 AM, Gregory Malsack <gmalsack at 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:6087294351 at 216.26.109.22:5060 SIP/2.0
>
> Call-ID: 31007e-31 at 147.135.32.221
>
> CSeq: 1 INVITE
>
> From: "Wi M"<sip:4144038968 at 147.135.32.221;user=phone>;tag=9bbc
>
> To: "Gregory Malsack"<sip:s at 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:4144038968 at 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-31 at 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:4144038968 at 147.135.32.221;user=phone>;tag=9bbc
>
> To: "Gregory Malsack"<sip:s at 216.26.109.22>;tag=as4fffe111
>
> Call-ID: 31007e-31 at 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-31 at 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:6087294351 at 216.26.109.22:5060 SIP/2.0
>
> Call-ID: 31007e-31 at 147.135.32.221
>
> CSeq: 1 ACK
>
> From: "Wi M"<sip:number from at 147.135.32.221;user=phone>;tag=9bbc
>
> To: "username"<sip:s at 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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