asterisk users - Mar 2007 - sip: failed the authenticate on INVITE

I've got a problem with a SIP Account I am trying to dial in with. The 
correct extension rings but when I pick up the call is not made and I 
get a busy signal. Dialing out works just fine - just calling this 
number doesn't seem to work.

Any pointers?

Thx
Michael

excerpt from sip.conf:
[general]
context=default
port = 5060           ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0    ; Address to bind to (all addresses on machine)
disable=all             ; Allow all codecs
allow=alaw
allow=ulaw
allow=g729
allow=gsm
srvlookup=yes
language=de

register => USERNAME:PWD@sip001.mitacs.com/mitacs_in_one
[...]


excerpt from extensions.conf
[global]
include => incoming
 [...]
[incoming]
exten => mitacs_in_one,1,Dial,SIP/100|25|r
exten => mitacs_in_one,2,Hangup


Debug from CLI:

    -- Executing [mitacs_in_one@incoming:1] 
Dial("SIP/213.XXX.XXX.XXX-40d08778", "SIP/100|25|r") in new
stack
    -- Called 100
    -- SIP/100-08231800 is ringing
    -- Call on SIP/100-08231800 left from hold
    -- SIP/100-08231800 answered SIP/213.185.164.125-40d08778
    -- Native bridging SIP/213.185.164.125-40d08778 and SIP/100-08231800
[Mar 29 10:14:20] NOTICE[9467]: chan_sip.c:11831 handle_response_invite: 
Failed to authenticate on INVITE to 
'<sip:43720NUMBER@sip001.mitacs.com>;tag=as2e46a760'
  == Spawn extension (incoming, mitacs_in_one, 1) exited non-zero on 
'SIP/213.XXX.XXX.XXX-40d08778'

Hi Michael,
have you tried to set canreinvite = no inside incoming calls context in 
sip.conf? Some SIP provider does not like reinvite.


Giorgio Incantalupo



Michael Zoller wrote:
> I've got a problem with a SIP Account I am trying to dial in with. The 
> correct extension rings but when I pick up the call is not made and I 
> get a busy signal. Dialing out works just fine - just calling this 
> number doesn't seem to work.
>
> Any pointers?
>
> Thx
> Michael
>
> excerpt from sip.conf:
> [general]
> context=default
> port = 5060           ; Port to bind to (SIP is 5060)
> bindaddr = 0.0.0.0    ; Address to bind to (all addresses on machine)
> disable=all             ; Allow all codecs
> allow=alaw
> allow=ulaw
> allow=g729
> allow=gsm
> srvlookup=yes
> language=de
>
> register => USERNAME:PWD@sip001.mitacs.com/mitacs_in_one
> [...]
>
>
> excerpt from extensions.conf
> [global]
> include => incoming
> [...]
> [incoming]
> exten => mitacs_in_one,1,Dial,SIP/100|25|r
> exten => mitacs_in_one,2,Hangup
>
>
> Debug from CLI:
>
>    -- Executing [mitacs_in_one@incoming:1] 
> Dial("SIP/213.XXX.XXX.XXX-40d08778", "SIP/100|25|r") in
new stack
>    -- Called 100
>    -- SIP/100-08231800 is ringing
>    -- Call on SIP/100-08231800 left from hold
>    -- SIP/100-08231800 answered SIP/213.185.164.125-40d08778
>    -- Native bridging SIP/213.185.164.125-40d08778 and SIP/100-08231800
> [Mar 29 10:14:20] NOTICE[9467]: chan_sip.c:11831 
> handle_response_invite: Failed to authenticate on INVITE to 
> '<sip:43720NUMBER@sip001.mitacs.com>;tag=as2e46a760'
>  == Spawn extension (incoming, mitacs_in_one, 1) exited non-zero on 
> 'SIP/213.XXX.XXX.XXX-40d08778'
>
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Giorgio Incantalupo wrote:
> have you tried to set canreinvite = no inside incoming calls context 
> in sip.conf? Some SIP provider does not like reinvite.
That seemed to have done the trick - Thank you!

Michael