inbound call issue…

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Asterisk Users 3 Comments

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@216.26.109.22:5060 SIP/2.0
Call-ID: href=”mailto:31007e-31@147.135.32.221″>31007e-31@147.135.32.221
CSeq: 1 INVITE
From: “Wi M”;tag=9bbc
To: “Gregory Malsack”
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:
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 – href=”mailto:31007e-31@147.135.32.221″>31007e-31@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”;tag=9bbc
To: “Gregory Malsack”;tag=as4fffe111
Call-ID: href=”mailto:31007e-31@147.135.32.221″>31007e-31@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@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@216.26.109.22:5060 SIP/2.0
Call-ID: href=”mailto:31007e-31@147.135.32.221″>31007e-31@147.135.32.221
CSeq: 1 ACK
From: “Wi M” href=”mailto:from@147.135.32.221″>from@147.135.32.221;user=phone>;tag=9bbc
To: “username”;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.broadvoice.com

[trunk_1]
type=peer
user=phone
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
fromuser=6087294351
secret=
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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