[Asterisk-Users] Strange Sip (FWD, SipGate and such) problem

[Thomas Gallaway]

Hi all

I use sipgate and FWD but seem not to get it going. I do not have NAT on 
the asterisk box (static ip).
The asterisk box has 2 network interfaces. One internal and one external.

Now when I make an call to a FWD or SipGate number all I get is
    -- Executing NoOp("SIP/113-6d2e", "") in new stack
    -- Executing Goto("SIP/113-6d2e", "intern-post|714551|1") in new stack
    -- Goto (intern-post,714551,1)
    -- Executing SetCallerID("SIP/113-6d2e", "270002") in new stack
    -- Executing SetCIDName("SIP/113-6d2e", "Thomas Gallaway") in new stack
    -- Executing Dial("SIP/113-6d2e", "SIP/14551 at fwd270002") in new stack
    -- Called 14551 at fwd270002
    -- SIP/fwd270002-6ee7 answered SIP/113-6d2e
    -- Attempting native bridge of SIP/113-6d2e and SIP/fwd270002-6ee7
  == Spawn extension (intern-post, 714551, 3) exited non-zero on 
'SIP/113-6d2e'

But either I get 1/2 second of audio or no audio. No matter how long I 
wait there is just no audio or just a short snippet of audio at the 
beginning.

Here is parts of my sip.conf;
[general]
port = 5060                     ; Port to bind to
localnet = 192.168.1.0         ; Internal NETWORK address
localmask = 255.255.255.0      ; Internal netmask
externip = 206.40.161.235
context = intern                ; Default for incoming calls
maxexpirey=3600
defaultexpirey=300
disallow=all                    ; Disallow all codecsa
allow=gsm
allow=alaw
allow=ulaw
tos=reliability
register => xxx:xxx at sipgate.de/150
register => xxx:xxx at fwd.pulver.com/151


[sipgate1]
type=friend
username=xxx
secret=xxx
host=sipgate.de
fromuser=xxx
fromdomain=sipgate.net
nat=no
context=incoming-sipgate
canreinvite=yes

[fwd270002]
allow=ulaw
type=friend
context=incoming-fwd
secret=xxx
username=xxx
host=fwd.pulver.com

Any ideas?
When I put nat=yes I actually will get 1 second of audio, then it dies.
I have been googling for a while now and not seem to find any sollution 
to this.

-- Thomas