[asterisk-users] transferring SIP call: no voice

[sean darcy]

I'm trying to connect a sip call from sipgate to Asterisk A to Asterisk 
B. Both are behind NAT, but port forwarded. I get the connection, but no 
voice - either in or out.

I can call on SIP from A to B (and from B to A). Do it all the time.

Asterisk A receives SIP calls from Junction and Teliax.

CLI on A looks right:
   == Using SIP RTP TOS bits 184
   == Using SIP RTP CoS mark 5
   == Using SIP VRTP CoS mark 6
   == Using UDPTL TOS bits 184
   == Using UDPTL CoS mark 5
     -- Executing [3008384e0 at sipgate-test:1] 
Answer("SIP/sipgate-00000016", "") in new stack
     -- Executing [3008384e0 at sipgate-test:2] 
Goto("SIP/sipgate-00000016", "home,447,1") in new stack
     -- Goto (home,447,1)
     -- Executing [447 at home:1] NoOp("SIP/sipgate-00000016", "xxxxxxxxx") 
in new stack
     -- Executing [447 at home:2] NoOp("SIP/sipgate-00000016", 
""yyyyyyyyyyy" <xxxxxxxxx>") in new stack
     -- Executing [447 at home:3] Dial("SIP/sipgate-00000016", 
"SIP/nhi-riverside-sip") in new stack
   == Using SIP RTP TOS bits 184
   == Using SIP RTP CoS mark 5
   == Using SIP VRTP CoS mark 6
   == Using UDPTL TOS bits 184
   == Using UDPTL CoS mark 5
     -- Called nhi-riverside-sip
     -- SIP/nhi-riverside-sip-00000017 answered SIP/sipgate-00000016
     -- Packet2Packet bridging SIP/sipgate-00000016 and 
SIP/nhi-riverside-sip-00000017

And on B:

     -- Executing [s at incoming:1] 
Answer("SIP/nhi-riverside-sip-00000009", "") in new stack
     -- Executing [s at incoming:2] NoOp("SIP/nhi-riverside-sip-00000009", 
"" callerid: ""yyyyyyyyyyyyy" <xxxxxxxxxxxxx>") in new stack
     -- Executing [s at incoming:3] Dial("SIP/nhi-riverside-sip-00000009", 
"DAHDI/g0,60") in new stack
     -- Called g0
     -- DAHDI/1-1 is ringing

Asterisk A sip.conf:

[sipgate]
type=friend
secret=  ;;SIP_PASSWORD
insecure=port,invite
defaultuser=  ;; SIP-ID
fromuser=      ;;SIP-ID
context=sipgate-test
fromdomain=sipgate.com
host=sipgate.com
outboundproxy=proxy.live.sipgate.com
qualify=yes
disallow=all
allow=ulaw
dtmfmode=rfc2833
nat=yes
canreinvite=no

Asterisk A extensions.conf:

[sipgate-test]
exten => _X.,1,Answer()
exten => _X.,n,GoTo(home,447,1)

[home]
exten =>447,1,NoOp(${CALLERID(num)})
exten =>447,n,NoOp(${CALLERID(all)})
exten=>447,n,Dial(SIP/nhi-riverside-sip)

And iptables on the router for Asterisk A:

$IPT -t nat -A PREROUTING -i $EXTIF  -p udp --dport 5060 -j DNAT --to 
10.10.10.180:5060
$IPT -A FORWARD -p udp --dport 5060 -m state --state NEW -d 10.10.10.180 
-j ACCEPT

# for sip, also port forward rtp ports
$IPT -t nat -A PREROUTING -i $EXTIF -p udp --dport 10000:20000 -j DNAT 
--to 10.10.11.180 # sip rtp
$IPT -A FORWARD -i $EXTIF -p udp --dport 10000:20000 -j ACCEPT

What am I missing?

sean