[asterisk-users] inbound call issue...

[Darrick Hartman]

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: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at 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<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
>>

>


-- 
Darrick Hartman
DJH Solutions, LLC
http://www.djhsolutions.com