Failed to authenticate on INVITE

Asterisk Asterisk Support
1

So I’ve been trolling forums and messing with this for 3 days now, to no avail.

NOTICE[1327]: chan_sip.c:13434 handle_response_invite: Failed to authenticate on INVITE to ’ “555*******” sip:192.168.x.xxx;tag=as03957a65.

Below is my dial plan, the secret and host are correct, anything else may be fairgame. Thanks in advance!

sip.conf

[general]
[1000]
type=peer
context=phones
host=dynamic
allowoverlap=no
bindport=5060
bindaddr=0.0.0.0
srvlookup=no
register=> 555*******:SecretPassword@sip3.voipvoip.com/555*******

[555*******]
type=peer
host=sip3.voipvoip.com
fromuser=555*******
secret=SecretPassword
context=incoming_calls
dtmfmode=rfc2833
disallow=all
allow=gsm
allow=ulaw
insecure=invite
deny=0.0.0.0/0
permit=sip3.voipvoip.com

extensions.conf

[globals]
[general]
[default]
exten => s,1,Verbose(1|Unrouted call handler)
exten => s,n,Answer()
exten => s,n,Wait(1)
exten => s,n,Hangup()
[incoming_calls]
exten => _X.,1,NoOp()
exten => _X.,n,Dial(SIP/1000)
exten => _X.,1,NoOp()
exten => _X.,n,Dial(SIP/sip3.voipvoip.com/${EXTEN})
[internal]
exten => 1000,1,Verbose(1|Extension 1000)
exten => 1000,n,Dial(SIP/1000,30)
exten => 1000,n,Hangup()
exten => 500,1,Verbose(1|Echo test application)
exten => 500,n,Echo()
exten => 500,nHangup()
[phones]
include => internal
include => outgoing_calls

Thanks,
Joe

2

Hi Joe,
do you have a sip debug of a call ?
thx

3

This is the message I get when I set debug:

[Mar 16 08:50:45] NOTICE[1327]: chan_sip.c:13434 handle_response_invite: Failed to authenticate on INVITE to ‘“555*******” sip:1000@12.168.67.107;tag=as22371c03’

<—Reliably Transmitting (no NAT) to 192.168.67.103:5060 —>
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.67.103:5060;branch=z9hG4bK-6c5fe7eb;received=192.168.67.103
From: “555*******1” sip:1000@192.168.67.107;tag=as2dd37cda
Call-ID:25cc583f-ac742676@192.168.67.103
CSeq: 101 INVITE
User-Agent"Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
X-Asterisk-HangupCause: Call Rejected
X-ASterisk-HangupCauseCode: 21
Content-Length: 0

<----------->

<— SIP read from 192.168.67.103:5060 —>
ACK sip:15406214588@192.168.67.107 SIP/2.0
Via: SIP/2.0/UDP 192.168.67.103:5060;branch=z9hG4bK-6c5fe7eb
From: “555*******” sip.1000@192.168.67.107;tag=5c1c55abf40067fao0
To: sip:1540621****@192.168.67.107;tag=as2dd37cda
Call-ID: 25cc583f-ac742676@192.168.67.103
CSeq: 101 ACK
Max-Forwards: 70
Contact: “555*******” sip:1000@192.168.67.103:5060
User-Agent: Linksys/SPA942-5.1.15(a)
Content-Length: 0

<----------->
— (10 headers 0 lines) —
Really destroying SIP dialog ‘7057574409bcb43318da176d491cb7@192.168.67.107’ Method: INVITE
Really destroying SIP dialog ‘25cc583f-ac742676@192.168.67.103’ Method: ACK

4

Were you able to resolve this issue? I’m having a similar error message come up on my setup and I have not been able to address the cause yet.

5

That’s a partial trace of the incoming call. The important information is in the trace of the outgoing call. Note that the trace will probably contain recoverable passwords.

6

I fixed by problem. For my SIP provided, I had to have username@ipaddress for the login authentication instead of ${EXTEN}@ipaddress. Once I did this, everything works fine.