asterisk users - Oct 2006 - Call from internal num. to VoIP gate

Greetings to All!

Help to solve a problem:

There is an asterisk and two VoIP a sluice (NSGate 800 2FXS 2FXO, NSGate 
800 2FXS).
In sip.conf they are registered so:

[3301]
type=friend
host=172.222.135.11
username=3301
secret=0000
defaultip=172.222.135.11
dtmfmode=rfc2833
context=it
callerid="VoIPGate2Line1" <3301>
allow=g723.1

[3302]
type=friend
host=172.222.135.11
username=3302
secret=0000
defaultip=172.222.135.11
dtmfmode=rfc2833
context=it
callerid="VoIPGate2Line2" <3302>
allow=g723.1

[3440]
type=friend
host=dynamic
username=3440
secret=0000
defaultip=10.11.11.10
dtmfmode=rfc2833
context=it
callerid="VoIPGateLine1" <3440>
allow=g723.1                   ; Asterisk only supports g723.1 pass-thru!

[3441]
type=friend
host=dynamic
username=3441
secret=0000
defaultip=10.11.11.10
dtmfmode=rfc2833
context=it
callerid="VoIPGateLine2" <3441>
allow=g723.1

[3442]
type=friend
host=dynamic
username=3442
secret=0000
defaultip=10.11.11.10
dtmfmode=rfc2833
context=it
callerid="VoIPGateLine3" <3442>
allow=g723.1

[3443]
type=friend
host=dynamic
username=3443
secret=0000
defaultip=10.11.11.10
dtmfmode=rfc2833
context=it
callerid="VoIPGateLine4" <3443>
allow=g723.1

When I call from internal telephone to some of this numbers  - the call 
goes, but when I pickup phone - communication breaks...
Please, help to understand!

Here a log:

<-- SIP read from 172.222.135.11:5060:
SIP/2.0 200 OK
From: "Unknown"<sip:Unknown@10.11.0.150>;tag=as2335e618
To: <sip:3301@172.222.135.11>;tag=ac16230b-13c4-ad-2c457-4966
Call-ID: 08001b0456b9377a479050064635a26e@10.11.0.150
CSeq: 102 INVITE
Via: SIP/2.0/UDP 10.11.0.150:5060;rport=5060;branch=z9hG4bK0e38ec58
Supported: replaces
User-Agent: FXS_GW (2sipfxs.112)
Contact: <sip:3301@172.222.135.11:0>
Content-Type: application/sdp
Content-Length: 220

v=0
o=FXS_GW 12367 0 IN IP4 172.222.135.11
s=Audio Session
i=Audio Session
c=IN IP4 172.222.135.11
t=0 0
m=audio 0 RTP/AVP 4 101
a=ptime:30
a=fmtp:101 0-11
a=rtpmap:4 G723/8000/1
a=rtpmap:101 telephone-event/8000

--- (11 headers 11 lines)---
Found RTP audio format 4
Found RTP audio format 101
Peer video RTP is at port 172.222.135.11:65535
Found description format G723
Found description format telephone-event
Capabilities: us - 0x8010f (g723|gsm|ulaw|alaw|g729|h263), peer - 
audio=0x1 (g723)/video=0x0 (nothing), combined - 0x1 (g723)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 
(telephone-event), combined - 0x1 (telephone-event)
set_destination: Parsing <sip:3301@172.222.135.11:0> for address/port to 
send to
set_destination: set destination to 172.222.135.11, port 0
Transmitting (no NAT) to 172.222.135.11:0:
ACK sip:3301@172.222.135.11:0 SIP/2.0
Via: SIP/2.0/UDP 10.11.0.150:5060;branch=z9hG4bK3f42da36;rport
From: "Unknown" <sip:Unknown@10.11.0.150>;tag=as2335e618
To: <sip:3301@172.222.135.11>;tag=ac16230b-13c4-ad-2c457-4966
Contact: <sip:Unknown@10.11.0.150>
Call-ID: 08001b0456b9377a479050064635a26e@10.11.0.150
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
Oct 30 08:15:34 WARNING[19154]: chan_sip.c:1082 __sip_xmit: sip_xmit of 
0x6f39d3c8 (len 387) to 172.222.135.11:0 returned -1: Invalid argument
Retransmitting #4 (no NAT) to 172.222.135.11:0:
BYE sip:3301@172.222.135.11:0 SIP/2.0
Via: SIP/2.0/UDP 10.11.0.150:5060;branch=z9hG4bK4ee204c1;rport
From: "Unknown" <sip:Unknown@10.11.0.150>;tag=as2335e618
To: <sip:3301@172.222.135.11>;tag=ac16230b-13c4-ad-2c457-4966
Contact: <sip:Unknown@10.11.0.150>
Call-ID: 08001b0456b9377a479050064635a26e@10.11.0.150
CSeq: 103 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
Oct 30 08:15:35 WARNING[19154]: chan_sip.c:1082 __sip_xmit: sip_xmit of 
0x8151630 (len 387) to 172.222.135.11:0 returned -1: Invalid argument
Destroying call 'C9792F0B-5BFF-4AB7-A75E-B56ABEB997BA@10.0.13.1'

<-- SIP read from 172.222.135.11:5060:
BYE sip:Unknown@10.11.0.150 SIP/2.0
From: <sip:3301@172.222.135.11>;tag=ac16230b-13c4-ad-2c457-4966
To: "Unknown"<sip:Unknown@10.11.0.150>;tag=as2335e618
Call-ID: 08001b0456b9377a479050064635a26e@10.11.0.150
CSeq: 1 BYE
Via: SIP/2.0/UDP 172.222.135.11:5060;branch=z9hG4bK-c0-30e94-d54
Max-Forwards: 70
Supported: replaces
User-Agent: FXS_GW (2sipfxs.112)
Content-Length: 0


--- (10 headers 0 lines)---
Sending to 172.222.135.11 : 5060 (non-NAT)
Transmitting (no NAT) to 172.222.135.11:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 
172.222.135.11:5060;branch=z9hG4bK-c0-30e94-d54;received=172.222.135.11
From: <sip:3301@172.222.135.11>;tag=ac16230b-13c4-ad-2c457-4966
To: "Unknown"<sip:Unknown@10.11.0.150>;tag=as2335e618
Call-ID: 08001b0456b9377a479050064635a26e@10.11.0.150
CSeq: 1 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:Unknown@10.11.0.150>
Content-Length: 0


---
Retransmitting #5 (no NAT) to 172.222.135.11:5060:
BYE sip:3301@172.222.135.11:0 SIP/2.0
Via: SIP/2.0/UDP 10.11.0.150:5060;branch=z9hG4bK4ee204c1;rport
From: "Unknown" <sip:Unknown@10.11.0.150>;tag=as2335e618
To: <sip:3301@172.222.135.11>;tag=ac16230b-13c4-ad-2c457-4966
Contact: <sip:Unknown@10.11.0.150>
Call-ID: 08001b0456b9377a479050064635a26e@10.11.0.150
CSeq: 103 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---

<-- SIP read from 172.222.135.11:5060:
SIP/2.0 481 Call Leg/Transaction Does Not Exist
From: "Unknown"<sip:Unknown@10.11.0.150>;tag=as2335e618
To: <sip:3301@172.222.135.11>;tag=ac16230b-13c4-ad-2c457-4966
Call-ID: 08001b0456b9377a479050064635a26e@10.11.0.150
CSeq: 103 BYE
Via: SIP/2.0/UDP 10.11.0.150:5060;rport=5060;branch=z9hG4bK4ee204c1
Supported: replaces
Content-Length: 0
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