Asterisk не посылает BYE

Доброго времени суток. Схема следующая: Аналоговые линии ->AddPack -> Asterisk -----интернет ----- SIP ATS --- какие-то телефоны

При звонке на аналоговую линии и отбое Астериск не посылает удаленной SIP ATS Panasonic BYE. Соответственно Panasonic не видит конца разговора и висит и шлет постоянно OPTIONS, на которые Астериск забивает и ничего не овечает

<--- SIP read from 77.51.yyy.yyy:35060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 94.158.xxx.xxx:5060;branch=z9hG4bK13392dfd;rport=5060
To: sip:000313@192.168.1.1;tag=22749
From: "asterisk" <sip:asterisk@94.158.xxx.xxx:5060>;tag=as31588114
Call-ID: 488facd267700ba25063068e3b1f5e2e@94.158.xxx.xxx
CSeq: 102 OPTIONS
Contact: sip:192.168.1.1:35060
Allow: INVITE,ACK,CANCEL,BYE,PRACK,OPTIONS,REGISTER,INFO,UPDATE
Accept-Encoding: *
Accept-Language: *
Server: Panasonic-MPR08-V4.2003/VSIPGW-V2.3001
Content-Length: 0


<------------->
--- (12 headers 0 lines) ---
Really destroying SIP dialog '488facd267700ba25063068e3b1f5e2e@94.158.xxx.xxx' Method: OPTIONS
Reliably Transmitting (NAT) to 77.51.yyy.yyy:35060:
OPTIONS sip:000312@192.168.1.1:35060 SIP/2.0
Via: SIP/2.0/UDP 94.158.xxx.xxx:5060;branch=z9hG4bK490f7675;rport
From: "asterisk" <sip:asterisk@94.158.xxx.xxx>;tag=as1398a4ae
To: <sip:000312@192.168.1.1:35060>
Contact: <sip:asterisk@94.158.xxx.xxx>
Call-ID: 047c11ee74192c7875341ee53b3a2535@94.158.xxx.xxx
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 23 Jan 2012 19:08:32 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0


---

<--- SIP read from 77.51.yyy.yyy:35060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 94.158.xxx.xxx:5060;branch=z9hG4bK490f7675;rport=5060
To: sip:000312@192.168.1.1;tag=26302
From: "asterisk" <sip:asterisk@94.158.xxx.xxx:5060>;tag=as1398a4ae
Call-ID: 047c11ee74192c7875341ee53b3a2535@94.158.xxx.xxx
CSeq: 102 OPTIONS
Contact: sip:192.168.1.1:35060
Allow: INVITE,ACK,CANCEL,BYE,PRACK,OPTIONS,REGISTER,INFO,UPDATE
Accept-Encoding: *
Accept-Language: *
Server: Panasonic-MPR08-V4.2003/VSIPGW-V2.3001
Content-Length: 0


<------------->
--- (12 headers 0 lines) ---
Really destroying SIP dialog '047c11ee74192c7875341ee53b3a2535@94.158.xxx.xxx' Method: OPTIONS
Reliably Transmitting (NAT) to 77.51.yyy.yyy:35060:
OPTIONS sip:000311@192.168.1.1:35060 SIP/2.0
Via: SIP/2.0/UDP 94.158.xxx.xxx:5060;branch=z9hG4bK59faa57c;rport
From: "asterisk" <sip:asterisk@94.158.xxx.xxx>;tag=as1cda8959
To: <sip:000311@192.168.1.1:35060>
Contact: <sip:asterisk@94.158.xxx.xxx>
Call-ID: 2c4292336f359c25557e17650e10c533@94.158.xxx.xxx
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 23 Jan 2012 19:08:32 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0

exten => _000xxxxxxxxxxxxxx,1,Answer()

exten => _000xxxxxxxxxxxxxx,n,Dial(SIP/${EXTEN}@${EXTEN:0:6},60)

exten => _000xxxxxxxxxxxxxx,n,Hangup()

Куда копать? В этом деле не силен, но хочется разобраться.

UPD:

Настройки sip для одной из линий АддПака

[000301]
type=friend
secret=xxxxxxxx
qualify=yes
port=5060
pickupgroup=1
deny=0.0.0.0/0.0.0.0
permit=xxx.xxx.xxx.xxx/255.255.255.255
host=dynamic
dtmfmode=inband
dial=SIP/000301
context=leb22gw
canreinvite=no
callgroup=1
accountcode=
call-limit=100
t38pt_udptl = yes