все извилины изломал чтоб пробросить транзитный звонок из sip в pri

ОХО 710/085.001
sip-транк стоит с меткой приват
pri-транк стоит с меткой паблик

распределение звонков sip-транка идет через privat numbering plan
на внутренние звонки звонок через sip-транк проходит нормально
на транзитные звонки (sip-> ОХО -> pri) ОХО шлет 404 not found

"privat numbering plan" прописана строка  finction - secondary trunk / start - 5 / end - 5/ base - ARS / nmt - keep / priv - no

в ARS есть направление 5 через PRI c меткой паблик, направление рабочее




   -- Executing [596729@outgoing-ChelnyOK:1] Macro("SIP/17426-00001675", "call,SIP/chally/596729") in new stack
   -- Executing [s@macro-call:1] Dial("SIP/17426-00001675", "SIP/chally/596729,30,tT") in new stack
 == Using SIP RTP CoS mark 5
 == Using SIP VRTP CoS mark 6
Audio is at 10.0.1.116 port 14950
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.1.157:5060:
INVITE sip:596729@192.168.1.157 SIP/2.0
Via: SIP/2.0/UDP 10.0.1.116:5060;branch=z9hG4bK66832217;rport
Max-Forwards: 70
From: "OK Audi Nab.Chelny" <sip:asterisk@10.0.1.116>;tag=as092fc5b1
To: <sip:596729@192.168.1.157>
Contact: <sip:asterisk@10.0.1.116>
Call-ID: 6c5d1e7e2e5ee1a375e14273100e2571@10.0.1.116
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.0.26-FONCORE-r78
Date: Wed, 14 Mar 2012 04:35:04 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 270

v=0
o=root 552228198 552228198 IN IP4 10.0.1.116
s=Asterisk PBX 1.6.0.26-FONCORE-r78
c=IN IP4 10.0.1.116
t=0 0
m=audio 14950 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

---
   -- Called chally/596729
trixtrixbox*CLI>
<--- SIP read from UDP://192.168.1.157:5060 --->
SIP/2.0 100 Trying
To: <sip:596729@192.168.1.157>
From: "OK Audi Nab.Chelny" <sip:asterisk@10.0.1.116>;tag=as092fc5b1
Call-ID: 6c5d1e7e2e5ee1a375e14273100e2571@10.0.1.116
CSeq: 102 INVITE
Via: SIP/2.0/UDP 10.0.1.116:5060;received=10.0.1.116;branch=z9hG4bK66832217;rport=5060
Content-Length: 0


<------------->
--- (7 headers 0 lines) ---
trixtrixbox*CLI>
<--- SIP read from UDP://192.168.1.157:5060 --->
SIP/2.0 404 Not Found
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, PRACK, REFER, NOTIFY, UPDATE
User-Agent: OxO_GW_710/142.001
To: <sip:596729@192.168.1.157>;tag=44e1bbc106bd77cd44eca0b175f4b12e
From: "OK Audi Nab.Chelny" <sip:asterisk@10.0.1.116>;tag=as092fc5b1
Call-ID: 6c5d1e7e2e5ee1a375e14273100e2571@10.0.1.116
CSeq: 102 INVITE
Via: SIP/2.0/UDP 10.0.1.116:5060;received=10.0.1.116;branch=z9hG4bK66832217;rport=5060
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---
Transmitting (no NAT) to 192.168.1.157:5060:
ACK sip:596729@192.168.1.157 SIP/2.0
Via: SIP/2.0/UDP 10.0.1.116:5060;branch=z9hG4bK66832217;rport
Max-Forwards: 70
From: "OK Audi Nab.Chelny" <sip:asterisk@10.0.1.116>;tag=as092fc5b1
To: <sip:596729@192.168.1.157>;tag=44e1bbc106bd77cd44eca0b175f4b12e
Contact: <sip:asterisk@10.0.1.116>
Call-ID: 6c5d1e7e2e5ee1a375e14273100e2571@10.0.1.116
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.0.26-FONCORE-r78
Content-Length: 0


---
   -- SIP/chally-00001676 is circuit-busy
 == Everyone is busy/congested at this time (1:0/1/0)
   -- Executing [s@macro-call:2] Goto("SIP/17426-00001675", "messages,CONGESTION,1") in new stack
   -- Goto (messages,CONGESTION,1)
 == Channel 'SIP/17426-00001675' jumping out of macro 'call'
   -- Sent into invalid extension 'CONGESTION' in context 'messages' on SIP/17426-00001675
   -- Executing [i@messages:1] Playback("SIP/17426-00001675", "/var/lib/asterisk/sounds/ru/ru/an-error-has-occured") in new stack
Really destroying SIP dialog '6c5d1e7e2e5ee1a375e14273100e2571@10.0.1.116' Method: INVITE
   -- <SIP/17426-00001675> Playing '/var/lib/asterisk/sounds/ru/ru/an-error-has-occured.gsm' (language 'en')=