односторонняя слышимость при исходящих

Всем привет! Имеется FreePBX 2.10 (астер 1.8).

eth0: 172.17.10.120 (внутреняя локалка), от нее выход в инет через НАТ (ip: xx.xx.xx.xx)

Проблема в односторонней слышимости. Т.е. кому я звоню меня слышит, а я его нет.

Возможно проблема в НАТ, но не пойму как исправить.

RTP debug показывает, что пакеты от меня уходят, а ко мне ничего не приходит.

настройка пира:

type=friend
fromuser=7777XXXXXXX
fromdomain=xx.xx.xx.xx
host=xx.xx.xx.xx
context=from-trunk
dtfmmode=rfc2833
disallow=all
allow=alaw&ulaw
insecure=invite
nat=yes
qualify=no

Sip Debag:

[2013-07-31 16:23:29] VERBOSE[25942] chan_sip.c: Reliably Transmitting (NAT) to yy.yy.yy.yy:5060:
INVITE sip:7701XXXXXXX@yy.yy.yy.yy SIP/2.0 
Via: SIP/2.0/UDP 172.17.10.120:5060;branch=z9hG4bK02d27ce7;rport 
Max-Forwards: 70 
From: "7777XXXXXXX" <sip:7777XXXXXXX@yy.yy.yy.yy>;tag=as2c4e4e14 
To: <sip:7701XXXXXXX@yy.yy.yy.yy> 
Contact: <sip:7777XXXXXXX@172.17.10.120:5060> 
Call-ID: 28f33b6a5ffacb7a77389c3855eaee01@yy.yy.yy.yy 
CSeq: 102 INVITE 
User-Agent: FPBX-2.10.1(1.8.20.1) 
Date: Wed, 31 Jul 2013 10:23:29 GMT 
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH 
Supported: replaces, timer 
Content-Type: application/sdp 
Content-Length: 263 

v=0 
o=root 1996754704 1996754704 IN IP4 172.17.10.120 
s=Asterisk PBX 1.8.20.1 
c=IN IP4 172.17.10.120 
t=0 0 
m=audio 12952 RTP/AVP 8 0 101 
a=rtpmap:8 PCMA/8000 
a=rtpmap:0 PCMU/8000 
a=rtpmap:101 telephone-event/8000 
a=fmtp:101 0-16 
a=ptime:20 
a=sendrecv 


[2013-07-31 16:23:30] VERBOSE[1692] chan_sip.c: Retransmitting #1 (NAT) to yy.yy.yy.yy:5060:
INVITE sip:7701XXXXXXX@yy.yy.yy.yy SIP/2.0 
Via: SIP/2.0/UDP 172.17.10.120:5060;branch=z9hG4bK02d27ce7;rport 
Max-Forwards: 70 
From: "7777XXXXXXX" <sip:7777XXXXXXX@yy.yy.yy.yy>;tag=as2c4e4e14 
To: <sip:7701XXXXXXX@yy.yy.yy.yy> 
Contact: <sip:7777XXXXXXX@172.17.10.120:5060> 
Call-ID: 28f33b6a5ffacb7a77389c3855eaee01@yy.yy.yy.yy 
CSeq: 102 INVITE 
User-Agent: FPBX-2.10.1(1.8.20.1) 
Date: Wed, 31 Jul 2013 10:23:29 GMT 
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH 
Supported: replaces, timer 
Content-Type: application/sdp 
Content-Length: 263 

v=0 
o=root 1996754704 1996754704 IN IP4 172.17.10.120 
s=Asterisk PBX 1.8.20.1 
c=IN IP4 172.17.10.120 
t=0 0 
m=audio 12952 RTP/AVP 8 0 101 
a=rtpmap:8 PCMA/8000 
a=rtpmap:0 PCMU/8000 
a=rtpmap:101 telephone-event/8000 
a=fmtp:101 0-16 
a=ptime:20 
a=sendrecv 

---
[2013-07-31 16:23:30] VERBOSE[1692] chan_sip.c: 
<--- SIP read from UDP:yy.yy.yy.yy:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.17.10.120:5060;received=xx.xx.xx.xx;branch=z9hG4bK02d27ce7;rport=63234
From: "7777XXXXXXX" <sip:7777XXXXXXX@yy.yy.yy.yy>;tag=as2c4e4e14
To: <sip:7701XXXXXXX@yy.yy.yy.yy>
Call-ID: 28f33b6a5ffacb7a77389c3855eaee01@yy.yy.yy.yy
CSeq: 102 INVITE

<------------->
[2013-07-31 16:23:30] VERBOSE[1692] chan_sip.c: --- (6 headers 0 lines) ---
[2013-07-31 16:23:32] VERBOSE[1692] chan_sip.c: 
<--- SIP read from UDP:yy.yy.yy.yy:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.17.10.120:5060;received=xx.xx.xx.xx;branch=z9hG4bK02d27ce7;rport=63234
From: "7777XXXXXXX" <sip:7777XXXXXXX@yy.yy.yy.yy>;tag=as2c4e4e14
To: <sip:7701XXXXXXX@yy.yy.yy.yy>;tag=oo00pll0-CC-33
Call-ID: 28f33b6a5ffacb7a77389c3855eaee01@yy.yy.yy.yy
CSeq: 102 INVITE
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,NOTIFY,MESSAGE
Contact: <sip:7701XXXXXXX@yy.yy.yy.yy:5060;transport=udp>
Content-Length: 207
Content-Type: application/sdp

v=0
o=HuaweiSoftx3000 1073743525 1073743525 IN IP4 yy.yy.yy.yy
s=SipCall
c=IN IP4 yy.yy.yy.yy
t=0 0
m=audio 18956 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=ptime:20
<------------->
[2013-07-31 16:23:32] VERBOSE[1692] chan_sip.c: --- (10 headers 9 lines) ---
[2013-07-31 16:23:32] VERBOSE[1692] chan_sip.c: list_route: hop: <sip:7701XXXXXXX@yy.yy.yy.yy:5060;transport=udp>
[2013-07-31 16:23:32] VERBOSE[1692] chan_sip.c: Found RTP audio format 8
[2013-07-31 16:23:32] VERBOSE[1692] chan_sip.c: Found RTP audio format 101
[2013-07-31 16:23:32] VERBOSE[1692] chan_sip.c: Found audio description format PCMA for ID 8
[2013-07-31 16:23:32] VERBOSE[1692] chan_sip.c: Found audio description format telephone-event for ID 101
[2013-07-31 16:23:32] VERBOSE[1692] chan_sip.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
[2013-07-31 16:23:32] VERBOSE[1692] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[2013-07-31 16:23:32] VERBOSE[1692] chan_sip.c: Peer audio RTP is at port yy.yy.yy.yy:18956
[2013-07-31 16:23:34] VERBOSE[1692] chan_sip.c: 
<--- SIP read from UDP:yy.yy.yy.yy:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 172.17.10.120:5060;received=xx.xx.xx.xx;branch=z9hG4bK02d27ce7;rport=63234
From: "7777XXXXXXX" <sip:7777XXXXXXX@yy.yy.yy.yy>;tag=as2c4e4e14
To: <sip:7701XXXXXXX@yy.yy.yy.yy>;tag=oo00pll0-CC-33
Call-ID: 28f33b6a5ffacb7a77389c3855eaee01@yy.yy.yy.yy
CSeq: 102 INVITE
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,NOTIFY,MESSAGE
Contact: <sip:7701XXXXXXX@yy.yy.yy.yy:5060;transport=udp>
P-Early-Media: sendrecv
Content-Length: 0