Исходящие SIP работают через ${EXTEN}, а 4444@prov.com - ошибка 484

Доброго времени суток! Прошу помощи более опытных товарищей. Собственно, исходящие SIP звонки работают через ${EXTEN}, а 4444@prov.com - ошибка 484 - Address incomplete. Тоисть, работает:

exten=>777,n,Dial(SIP/zadarma/${EXTEN},30,m)

не работает:

exten=>777,n,Dial(SIP/zadarma/10005@ipshka.com,30,m)

Asterisk 1.8 за NATом, в sip.conf nat=yes . Немогу понять в чем проблема и куда копать.

Вот кусок дебага при звонке:

   -- Called SIP/zadarma/100005@ipshka.com
    -- Started music on hold, class 'default', on SIP/1002-00000015
Audio is at 16792
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

--- Transmitting (NAT) to 192.168.1.4:5060 ---
SIP/2.0 183 Session Progress

Заранее благодарю за помощь! Via: SIP/2.0/UDP 192.168.1.4;branch=z9hG4bKswhlnhdg;received=192.168.1.4;rport=5060 From: "1002" <sip:1002@192.168.1.10>;tag=tocdx To: <sip:777@192.168.1.10>;tag=as15640573 Call-ID: gkpbqafmvcgrpys@mars18 CSeq: 548 INVITE Server: Asterisk PBX 1.8.15-cert2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: <sip:777@192.168.1.10:5060> Content-Type: application/sdp Content-Length: 266

v=0
o=root 686847662 686847662 IN IP4 192.168.1.10
s=Asterisk PBX 1.8.15-cert2
c=IN IP4 192.168.1.10
t=0 0
m=audio 16792 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

------------

<--- SIP read from UDP:144.76.42.108:5060 --->
SIP/2.0 484 Address Incomplete
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK1e5e5d6e;rport=5060;received=188.0.65.19
From: "1002" <sip:44665@sip.zadarma.com>;tag=as51186224
To: <sip:100005@ipshka.com@sipde.zadarma.com>;tag=b638310eda6e4a73cf10b7fe3c94c572.8320
Call-ID: 533589d31603074d17e86ffc3f4f10a5@sip.zadarma.com
CSeq: 102 INVITE
Server: kamailio (4.1.2 (x86_64/linux))
Content-Length: 0

-------------
--- (8 headers 0 lines) ---
    -- Got SIP response 484 "Address Incomplete" back from 144.76.42.108:5060
Transmitting (NAT) to 144.76.42.108:5060:
ACK sip:100005@ipshka.com@sipde.zadarma.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK1e5e5d6e;rport
Max-Forwards: 70
From: "1002" <sip:44665@sip.zadarma.com>;tag=as51186224
To: <sip:100005@ipshka.com@sipde.zadarma.com>;tag=b638310eda6e4a73cf10b7fe3c94c572.8320
Contact: <sip:44665@192.168.1.10:5060>
Call-ID: 533589d31603074d17e86ffc3f4f10a5@sip.zadarma.com
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.15-cert2
Content-Length: 0


---
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Stopped music on hold on SIP/1002-00000015
    -- Executing [777@sip-incoming:3] Hangup("SIP/1002-00000015", "") in new stack
  == Spawn extension (sip-incoming, 777, 3) exited non-zero on 'SIP/1002-00000015'
Scheduling destruction of SIP dialog 'gkpbqafmvcgrpys@mars18' in 32000 ms (Method: INVITE)

p:1002@192.168.1.10>;tag=tocdx
To: <sip:777@192.168.1.10>;tag=as15640573
Call-ID: gkpbqafmvcgrpys@mars18
CSeq: 548 INVITE
Server: Asterisk PBX 1.8.15-cert2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


------------
Really destroying SIP dialog '533589d31603074d17e86ffc3f4f10a5@sip.zadarma.com' Method: INVITE