2 same sip extension number on 2 asterisk – call not passing on certain condition

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Hi list,

something crazy here. 2 asterisk on 2 different place (1.4 and 1.8) both
having an extension [115], one as type peer (caller side 1.4) and one as
friend (callee side 1.8). Phones from both location connect to Asterisk
from LAN. Router are Linux boxes.

Connection between the 2 sites is done like this:

On the callee side

[115] ;callee
type=friend
host=dynamic
secret=otherSecret
context=local
nat=no
canreinvite=no
qualify=no
dtmfmode=rfc2833
allow=all
call-limit=1
busy-level=1
allow=all

[Caller]
type=peer
host=voip1.domain.net
deny=0.0.0.0/0.0.0.0
permit=xxx.xxx.xxx.xxx
context=myAccess
disallow=all
allow=all
nat=yes
insecure=port,invite

On the caller side

[115]
type=peer
username=115
secret=blabla
context=local
host=dynamic
nat=yes
canreinvite=no
dtmfmode=auto
disallow=all
allow=jpeg,png,h263,h263p,h264,alaw,ulaw
callgroup=1
pickupgroup=1
insecure=invite

[Callee]
type=peer
host=voip1.other-domain.net
deny=0.0.0.0/0.0.0.0
permit=yyy.yyy.yyy.yyy
context=myOtherAccess
disallow=all
allow=all

Now when I call from 115@caller to any number at callee side I’m
rejected with

Sending to xxx.xxx.xxx.xxx:5060 (no NAT)
Using INVITE request as basis request –
281799ed7524c46966bcf303371edba4@xxx.xxx.xxx.xxx
Found peer ‘115’ for ‘115’ from xxx.xxx.xxx.xxx:5060

< --- Reliably Transmitting (no NAT) to xxx.xxx.xxx.xxx:5060 --->
SIP/2.0 401 Unauthorized

This is, Asterisk try to authenticate on URI SIP user before from peer
definition. If I change type from friend to peer it worked (I need the
friend for this extension)

Does someone has an idea on how to solve this problem?

Thanks for any hint