Hi, I upgraded my two asterisk boxes today to the latest cvs (up from 5/3/04). These two boxes talk to eachother via sip, not iax. Since the upgrade, I get the error "Failed to authenticate on INVITE" trying to make calls to/from either box. Removing the secret from each box's sip config seems to work but is utterly braindead. Has anyone seen this? - Eric
At 16:49 16/06/2004 -0400, Eric wrote:>I upgraded my two asterisk boxes today to the latest cvs (up from 5/3/04). > >These two boxes talk to eachother via sip, not iax. Since the upgrade, I >get the error "Failed to authenticate on INVITE" trying to make calls to/from >either box. Removing the secret from each box's sip config seems to work but >is utterly braindead.include the line in sip.conf for each user the call insecure=yes ; To match a peer based by IP address only and not peer
NOTICE[98310]: chan_sip.c:6638 handle_response: Failed to authenticate on INVITE
to 'sip:1234@mysipprovider.com;tag=as0f1d3429'
sip.conf
------------
register => 1234:password@mysipprovider.com
------------
extension.conf
--------------
;
; Own extensions
;
exten => 0852509516,1,Goto(resepsjon-own,s,1)
;
[resepsjon-own]
;
exten => s,1,Answer
exten => s,2,SetMusicOnHold(default)
exten => s,3,DigitTimeout,5
exten => s,4,ResponseTimeout,15
exten => s,5,Background(own/choose) ; Meny,
1 for support, 2 for support, 3 for wx3
exten => s,6,Wait(1)
exten => s,7,Background(own/choosenumber) ; dialer
pushes a # ,and being sent to..
;
ip-phone must be picked up in ,20000ms,tr or hangup
exten => 1,1,Goto(privatanslutningar,s,1)
exten => 2,1,Goto(foretagsanslutningar,s,1)
; #=hangup
exten => #,1,Playback(custom/no-key-registered)
exten => #,2,Hangup
exten => t,1,Goto(#,1) ; If they take too long, give up
exten => i,1,Playback(invalid) ; "That's not valid, try again"
inmenu]
;
[privatanslutningar]
;
exten => s,1,Answer
exten => s,2,SetMusicOnHold(default)
exten => s,3,DigitTimeout,5
exten => s,4,ResponseTimeout,15
exten => s,5,Background(own/privatanslutningar) ; Meny, 1 for
support, 2 for support, 3 for wx3
; dialer pushes a # ,and being sent to..
exten => 1,1,Answer
exten => 1,2,Queue(help-privatanslutningar-queue)
exten => 2,1,Answer
exten => 2,2,Queue(order-privatanslutningar-queue)
exten => 3,1,Answer
exten => 3,2,Queue(info-privatanslutningar-queue)
; #=hangup
;exten => #,1,Playback(custom/no-key-registered)
;exten => #,2,Hangup
exten => t,1,Queue(general-privatanslutningar-queue) ; If they
take too long, give up
exten => i,1,Playback(invalid)
; "That's not valid, try again" inmenu]
;
[foretagsanslutningar]
;
exten => s,1,Answer
exten => s,2,SetMusicOnHold(default)
exten => s,3,DigitTimeout,5
exten => s,4,ResponseTimeout,15
exten => s,5,Background(own/foretagsanslutningar) ; Meny, 1 for
support, 2 for support, 3 for wx3
; dialer pushes
a # ,and being sent to..
; ip-phone must
be picked up in ,20000ms,tr or hangup
exten => 1,1,Answer
exten => 1,2,Queue(info-bedriftsanslutningar-queue)
exten => 2,1,Answer
exten => 2,2,Queue(help-bedriftsanslutningar-queue)
exten => 3,1,Answer
exten => 3,2,Queue(error-bedriftsanslutningar-queue)
; #=hangup
;exten => #,1,Playback(custom/no-key-registered)
;exten => #,2,Hangup
exten => t,1,Queue(general-bedriftsanslutningar-queue) ; If
they take too long, give up
exten => i,1,Playback(invalid) ;
"That's not valid, try again" inmenu]
--------------
The call gets into queue, then... the other phone rings.. and when I pick up - I
get this message:
NOTICE[98310]: chan_sip.c:6638 handle_response: Failed to authenticate on INVITE
to 'sip:1234@mysipprovider.com;tag=as0f1d3429'
I know that the register => works.. I have checked with my SIP-provider, and
they say that it is logged in.
What else can be wrong ?
/ Stig Henning
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20040917/b9feb60e/attachment.htm
I am getting this also.
I am trying to get Asterisk to talk similarly to BT Communicator to the BT
server. I can register but then the INVITE fails.
BT are mixed up with their domains (in fact in the INVITE their software has
a To: header with <number>@domain1 and an auth URI referencing
<number>@domain2. The realm is domain1.) This can't be done in
Asterisk
where it is consistent about the URI.
I had been blaming this, but if you are having problems too...
I get the standard 407 header requesting Proxy Auth for the call. Asterisk
submits the INVITE with auth and after the usual "Trying" I just get
another
407. I have traces of Asterisk and the client which works and they seem so
similar in what they do. I have made all the port ranges the same too. BT
Communicator fails if you use port 5060 for the SIP client - they use 5052.
Peter
-----Original Message-----
From: Stig Thune [mailto:stig.thune@telecoms-resources.no]
Sent: 17 September 2004 12:55
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Failed to authenticate on INVITE
NOTICE[98310]: chan_sip.c:6638 handle_response: Failed to authenticate on
INVITE to 'sip:1234@mysipprovider.com;tag=as0f1d3429'
sip.conf
------------
register => 1234:password@mysipprovider.com
------------
extension.conf
--------------
;
; Own extensions
;
exten => 0852509516,1,Goto(resepsjon-own,s,1)
;
[resepsjon-own]
;
exten => s,1,Answer
exten => s,2,SetMusicOnHold(default)
exten => s,3,DigitTimeout,5
exten => s,4,ResponseTimeout,15
exten => s,5,Background(own/choose) ; Meny,
1 for support, 2 for support, 3 for wx3
exten => s,6,Wait(1)
exten => s,7,Background(own/choosenumber) ;
dialer pushes a # ,and being sent to..
;
ip-phone must be picked up in ,20000ms,tr or hangup
exten => 1,1,Goto(privatanslutningar,s,1)
exten => 2,1,Goto(foretagsanslutningar,s,1)
; #=hangup
exten => #,1,Playback(custom/no-key-registered)
exten => #,2,Hangup
exten => t,1,Goto(#,1) ; If they take too long, give up
exten => i,1,Playback(invalid) ; "That's not valid, try again"
inmenu]
;
[privatanslutningar]
;
exten => s,1,Answer
exten => s,2,SetMusicOnHold(default)
exten => s,3,DigitTimeout,5
exten => s,4,ResponseTimeout,15
exten => s,5,Background(own/privatanslutningar) ; Meny, 1 for
support, 2 for support, 3 for wx3
; dialer pushes a # ,and being sent to..
exten => 1,1,Answer
exten => 1,2,Queue(help-privatanslutningar-queue)
exten => 2,1,Answer
exten => 2,2,Queue(order-privatanslutningar-queue)
exten => 3,1,Answer
exten => 3,2,Queue(info-privatanslutningar-queue)
; #=hangup
;exten => #,1,Playback(custom/no-key-registered)
;exten => #,2,Hangup
exten => t,1,Queue(general-privatanslutningar-queue) ; If they
take too long, give up
exten => i,1,Playback(invalid)
; "That's not valid, try again" inmenu]
;
[foretagsanslutningar]
;
exten => s,1,Answer
exten => s,2,SetMusicOnHold(default)
exten => s,3,DigitTimeout,5
exten => s,4,ResponseTimeout,15
exten => s,5,Background(own/foretagsanslutningar) ; Meny, 1 for
support, 2 for support, 3 for wx3
; dialer
pushes a # ,and being sent to..
; ip-phone
must be picked up in ,20000ms,tr or hangup
exten => 1,1,Answer
exten => 1,2,Queue(info-bedriftsanslutningar-queue)
exten => 2,1,Answer
exten => 2,2,Queue(help-bedriftsanslutningar-queue)
exten => 3,1,Answer
exten => 3,2,Queue(error-bedriftsanslutningar-queue)
; #=hangup
;exten => #,1,Playback(custom/no-key-registered)
;exten => #,2,Hangup
exten => t,1,Queue(general-bedriftsanslutningar-queue) ; If
they take too long, give up
exten => i,1,Playback(invalid) ;
"That's
not valid, try again" inmenu]
--------------
The call gets into queue, then... the other phone rings.. and when I pick up
- I get this message:
NOTICE[98310]: chan_sip.c:6638 handle_response: Failed to authenticate on
INVITE to 'sip:1234@mysipprovider.com;tag=as0f1d3429'
I know that the register => works.. I have checked with my SIP-provider, and
they say that it is logged in.
What else can be wrong ?
/ Stig Henning
This e-mail and any attachment is for authorised use by the intended
recipient(s) only. It may contain proprietary material, confidential information
and/or be subject to legal privilege. It should not be copied, disclosed to,
retained or used by, any other party. If you are not an intended recipient then
please promptly delete this e-mail and any attachment and all copies and inform
the sender. Thank you.
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20040917/7e9e70ae/attachment.htm
For info
The new chan_sip2.c and recent CVS (yesterday) fix this and I can now use
Asterisk to make calls on the sip.btcommunicator.bt.net service. If anyone
wants help with the settings, e-mail me off list.
:)
Peter
-----Original Message-----
From: Whisker, Peter [mailto:Peter.Whisker@logicacmg.com]
Sent: 17 September 2004 14:40
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Failed to authenticate on INVITE
I am getting this also.
I am trying to get Asterisk to talk similarly to BT Communicator to the BT
server. I can register but then the INVITE fails.
BT are mixed up with their domains (in fact in the INVITE their software has
a To: header with <number>@domain1 and an auth URI referencing
<number>@domain2. The realm is domain1.) This can't be done in
Asterisk
where it is consistent about the URI.
I had been blaming this, but if you are having problems too...
I get the standard 407 header requesting Proxy Auth for the call. Asterisk
submits the INVITE with auth and after the usual "Trying" I just get
another
407. I have traces of Asterisk and the client which works and they seem so
similar in what they do. I have made all the port ranges the same too. BT
Communicator fails if you use port 5060 for the SIP client - they use 5052.
Peter
-----Original Message-----
From: Stig Thune [mailto:stig.thune@telecoms-resources.no]
Sent: 17 September 2004 12:55
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Failed to authenticate on INVITE
NOTICE[98310]: chan_sip.c:6638 handle_response: Failed to authenticate on
INVITE to 'sip:1234@mysipprovider.com;tag=as0f1d3429'
sip.conf
------------
register => 1234:password@mysipprovider.com
------------
extension.conf
--------------
;
; Own extensions
;
exten => 0852509516,1,Goto(resepsjon-own,s,1)
;
[resepsjon-own]
;
exten => s,1,Answer
exten => s,2,SetMusicOnHold(default)
exten => s,3,DigitTimeout,5
exten => s,4,ResponseTimeout,15
exten => s,5,Background(own/choose) ; Meny,
1 for support, 2 for support, 3 for wx3
exten => s,6,Wait(1)
exten => s,7,Background(own/choosenumber) ;
dialer pushes a # ,and being sent to..
;
ip-phone must be picked up in ,20000ms,tr or hangup
exten => 1,1,Goto(privatanslutningar,s,1)
exten => 2,1,Goto(foretagsanslutningar,s,1)
; #=hangup
exten => #,1,Playback(custom/no-key-registered)
exten => #,2,Hangup
exten => t,1,Goto(#,1) ; If they take too long, give up
exten => i,1,Playback(invalid) ; "That's not valid, try again"
inmenu]
;
[privatanslutningar]
;
exten => s,1,Answer
exten => s,2,SetMusicOnHold(default)
exten => s,3,DigitTimeout,5
exten => s,4,ResponseTimeout,15
exten => s,5,Background(own/privatanslutningar) ; Meny, 1 for
support, 2 for support, 3 for wx3
; dialer pushes a # ,and being sent to..
exten => 1,1,Answer
exten => 1,2,Queue(help-privatanslutningar-queue)
exten => 2,1,Answer
exten => 2,2,Queue(order-privatanslutningar-queue)
exten => 3,1,Answer
exten => 3,2,Queue(info-privatanslutningar-queue)
; #=hangup
;exten => #,1,Playback(custom/no-key-registered)
;exten => #,2,Hangup
exten => t,1,Queue(general-privatanslutningar-queue) ; If they
take too long, give up
exten => i,1,Playback(invalid)
; "That's not valid, try again" inmenu]
;
[foretagsanslutningar]
;
exten => s,1,Answer
exten => s,2,SetMusicOnHold(default)
exten => s,3,DigitTimeout,5
exten => s,4,ResponseTimeout,15
exten => s,5,Background(own/foretagsanslutningar) ; Meny, 1 for
support, 2 for support, 3 for wx3
; dialer
pushes a # ,and being sent to..
; ip-phone
must be picked up in ,20000ms,tr or hangup
exten => 1,1,Answer
exten => 1,2,Queue(info-bedriftsanslutningar-queue)
exten => 2,1,Answer
exten => 2,2,Queue(help-bedriftsanslutningar-queue)
exten => 3,1,Answer
exten => 3,2,Queue(error-bedriftsanslutningar-queue)
; #=hangup
;exten => #,1,Playback(custom/no-key-registered)
;exten => #,2,Hangup
exten => t,1,Queue(general-bedriftsanslutningar-queue) ; If
they take too long, give up
exten => i,1,Playback(invalid) ;
"That's
not valid, try again" inmenu]
--------------
The call gets into queue, then... the other phone rings.. and when I pick up
- I get this message:
NOTICE[98310]: chan_sip.c:6638 handle_response: Failed to authenticate on
INVITE to 'sip:1234@mysipprovider.com;tag=as0f1d3429'
I know that the register => works.. I have checked with my SIP-provider, and
they say that it is logged in.
What else can be wrong ?
/ Stig Henning
This e-mail and any attachment is for authorised use by the intended
recipient(s) only. It may contain proprietary material, confidential
information and/or be subject to legal privilege. It should not be copied,
disclosed to, retained or used by, any other party. If you are not an
intended recipient then please promptly delete this e-mail and any
attachment and all copies and inform the sender. Thank you.
This e-mail and any attachment is for authorised use by the intended
recipient(s) only. It may contain proprietary material, confidential information
and/or be subject to legal privilege. It should not be copied, disclosed to,
retained or used by, any other party. If you are not an intended recipient then
please promptly delete this e-mail and any attachment and all copies and inform
the sender. Thank you.
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20040921/f7beb955/attachment.htm
Esteban Barrientos Abarca
2004-Sep-30 13:05 UTC
[Asterisk-Users] Failed to authenticate on INVITE
Hi!
I have an asterisk server, it has been running for a few months and everything
is great. Now i need it working with a sip provider but i'm having some
problems.
When i try to make a phone call i get this message:
-- Executing Dial("SIP/esteban-aad5",
"SIP/XXXXXXXXXX@provider||rtT") in new stack
-- Called XXXXXXXXXX@provider
Sep 30 13:38:59 NOTICE[98311]: chan_sip.c:6557 handle_response: Failed to
authenticate on INVITE
to'"esteban"<sip:XXXXXXXXXX@sip.myprovider.com>;tag=as34b3dbce'
this is in my sip.conf file
[provider]
type=peer ; we only want to call out, not be called
secret=mysecret
username=XXXXXXXXXX
host=sip.myprovider.com
nat=no
canreinvite=no
fromuser=XXXXXXXXXX
fromdomain=sip.myprovider.com
context=sip
insecure=very
[esteban]
type=friend
host=dynamic
username=esteban
secret=mysecret
dtmfmode=RFC2833
mailbox=303
context=sip
esteban is the username of the phone i'm dialing from.
Am i missing any parameter???
Thanks.
Esteban