[asterisk-users] Websockets on Asterisk 11 and SipML5

[James Mortensen]

Joshua Colp <jcolp <at> digium.com> writes:

> 
> Hi James,
> 
> I've trimmed the thread down, well, completely. ^_^
> 
>  From looking at your information and reading the code it looks as 
> though there is a case where this may occur if certain NAT options are 
> enabled. This is certainly a bug as the code should just not execute 
> when WebSocket is involved. For an immediate fix you can set nat=no in 
> the entry in sip.conf. This should change the result and would also 
> explain why this has not been seen by others.
> 


Hi Joshua,

I'm still getting the same result. Here is what I have in my sip.conf:


[general]
context=default                 ; Default context for incoming calls
srvlookup=yes
port=5060
bindaddr=0.0.0.0
;pedantic=no
rtcachefriends=yes
dtmfmode=auto
disallow=all
allow=g729
allow=ulaw                     ; Allow codecs in order of preference
allow=ilbc
;allow=silk8
allow=gsm
;allow=silk16
;allow=silk24

;nat=force_rport
nat=no    
externip=example.org
localnet=10.168.151.65/255.255.254.0
qualify=yes


[3001]
username=3001
secret=xxxxx
host=dynamic
type=friend
context=test
transport=ws
nat=no

[3002]
username=3002
secret=xxxxx
host=dynamic
type=friend
context=test
transport=ws
nat=no