asterisk users - Mar 2011 - Trunk form asterisk1 to asterisk2 fails

Hello,

When I want to send a call from asterisk-server 1 to asterisk-server 2, 
it fails.

On Asterisk server 1 :

register => user:passwd at asterisk1 ; Test TRUNK
[trunk2]
type=peer
host=asterisk1
username=user
;defaultuser=user
secret=passwd
disallow=all
allow=alaw
allow=gsm
qualify=yes
canreinvite=no
dtmfmode=rfc2833

Dialplan on Asterisk server 1 :

exten => 3291,1,NoOp()
exten => 3291,n,Set(${CALLERID(all)}="3291" <3291>)
exten => 3291,n,Dial(SIP/trunk2/3291)


On Asterisk server 2 I see the following when I make a call with a 
Grandstream IP-phone, registered at Asterisk server 1 :

[Mar 16 20:32:44] WARNING[1680]: chan_sip.c:12872 check_auth: username 
mismatch, have <test7>, digest has <user>
[Mar 16 20:32:44] NOTICE[1680]: chan_sip.c:20235 handle_request_invite: 
Failed to authenticate device "T 7" <sip:test7 at
192.168.1.150>;tag=as5a2d92df



How come the credentials of the Grandstream IP-phone are used and not 
the username + password of [trunk2] ??


Kind regards,
Jonas.
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On 03/16/2011 08:39 PM, Jonas Kellens wrote:
>
> On Asterisk server 2 I see the following when I make a call with a 
> Grandstream IP-phone, registered at Asterisk server 1 :
>
> [Mar 16 20:32:44] WARNING[1680]: chan_sip.c:12872 check_auth: username 
> mismatch, have <test7>, digest has <user>
> [Mar 16 20:32:44] NOTICE[1680]: chan_sip.c:20235 
> handle_request_invite: Failed to authenticate device "T 7" 
> <sip:test7 at 192.168.1.150>;tag=as5a2d92df
>
>
>
> How come the credentials of the Grandstream IP-phone are used and not 
> the username + password of [trunk2] ??
Found the answer to my own question : fromuser in the peer definition


Kind regards,
Jonas.

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Jonas Kellens <jonas.kellens <at> telenet.be> writes:
> 
> 
> On 03/16/2011 08:39 PM, Jonas Kellens wrote:
> 
> 
> Found the answer to my own question : fromuser in the peer definition
> Kind regards,
> Jonas.
> 
> 
> --
> _____________________________________________________________________

Can you extend a little bit this fix? I have a similar problem forwarding a call
to another Asterisk. Thank you.