[asterisk-bugs] [JIRA] (ASTERISK-25372) SIP/2.0 401 Unauthorized for incoming calls.

[Rusty Newton (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-25372?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Rusty Newton updated ASTERISK-25372:
------------------------------------

    Assignee: Agasthian P
      Status: Waiting for Feedback  (was: Triage)

You shouldn't need insecure=invite,port to do what you want and that is probably very unsafe.

Your debug indicates that Asterisk is matching up and identifying the peer. So I believe if you remove the "secret" option for your peer then Asterisk should not require password authentication for that peer.

You should consult with the FreePBX community about the best way to do that. This appears to be a support issue and not a bug.

{noformat}
[2723]
deny=0.0.0.0/0.0.0.0
secret=7165316a8ba3e6de9b6d8bd28777f1bc
dtmfmode=rfc2833
canreinvite=no
context=from-internal
host=dynamic
trustrpid=yes
mediaencryption=no
sendrpid=pai
type=friend
nat=force_rport,comedia
port=5060
qualify=yes
qualifyfreq=60
transport=udp,tcp,tls
avpf=no
force_avp=no
icesupport=no
encryption=no
callgroup=
pickupgroup=
dial=SIP/2723
mailbox=2723 at default
permit=0.0.0.0/0.0.0.0
callerid=Agasthian <2723>
callcounter=yes
faxdetect=no
cc_monitor_policy=generic
{noformat}

> SIP/2.0 401 Unauthorized for incoming calls.
> --------------------------------------------
>
>                 Key: ASTERISK-25372
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-25372
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_sip/General
>    Affects Versions: 11.19.0
>         Environment: SHMZ release 6.5 (Final)
> Linux knaufsappdc 2.6.32-431.el6.x86_64 
> Running on a Virtual Environment. VMWare
>            Reporter: Agasthian P
>            Assignee: Agasthian P
>            Severity: Minor
>         Attachments: coretracing.txt, sipconffiles1, siptrace.txt
>
>
> Running Freepbx on top of Asterisk 11.19.0.
> SIP trunk between Cisco Call Manager and FreePbx Asterisk.
> All the incoming calls from the Cisco Call Manager through SIP trunk are getting the error "SIP/2.0 401 Unauthorized". The Cisco Call Manager SIP trunks are treated as peer to peer and do not need any kind of registration.
> Have checked all the settings required to not enable any kind of authentication, still it seems to be asking for authentication.
> Tried the forums first, and have taken care of the following settings :-
> 1. insecure=invite,port
> 2.qualify=yes
> 3. type=friend
> 4. allow sip guests= yes
> 5. Allow Anonymous Inbound SIP Calls=yes
> The sip logs and the sip configuration files are attached as files.
> Call Flow :-
> FreePBX(20.1.1.58) - CUCM(20.1.1.170)
> 2723 extension is in both the systems. And 2723(CUCM) is trying to call 2723(Asterisk).



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