Приветствую

Подняли на Cisco 2811  PRI E1.
Звонки проходят нормально со станции на астериск и обратно,

Но проблема в том, что при звонке со станции на астериск звонок приходит, говорим, ложим трубку, И, с этого момента начинаются повторные вызова с cisco на asterisk с интервалом 40-45сек.

Если с астериска звонить через cisco, повторов нет.

Схема:
Станция-Е1-Cisco2811-eth-Asterisk(482600)

Что не так, просветите, пожалуйста.

Debug Cisco2811:

!
version 12.4
service nagle
no service pad
service tcp-keepalives-in
service tcp-keepalives-out
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
!
boot-start-marker
boot-end-marker
!
card type e1 0 0
card type e1 0 1
logging buffered 51200 warnings
!
no aaa new-model
network-clock-participate wic 0
no network-clock-participate wic 1
!
!
ip cef
!
!
no ip domain lookup
isdn switch-type primary-net5
isdn voice-call-failure 0
!
voice-card 0
codec complexity high
no dspfarm
!
!
!
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
h323
sip
  registrar server expires max 3600 min 3600
  redirect contact order best-match
  no call service stop
!
!
!
voice class codec 1
codec preference 1 g723r53 bytes 40
codec preference 2 g729br8 bytes 30
!
voice class codec 2
codec preference 1 g729br8 bytes 30
codec preference 2 g711alaw
!
voice class codec 3
codec preference 1 g729r8
!
!
controller E1 0/0/0
framing NO-CRC4
pri-group timeslots 1-31
!        
!        
!        
translation-rule 60
Rule 1 600 482600
Rule 2 601 482601
Rule 3 602 482602
!        
!        
!        
!        
interface FastEthernet0/0
ip address 172.30.3.119 255.255.255.0
duplex auto
speed auto
!
interface Serial0/0/0:15
no ip address
encapsulation hdlc
isdn switch-type primary-net5
isdn overlap-receiving
isdn incoming-voice voice
no keepalive
no cdp enable
!
no ip forward-protocol nd
ip route 0.0.0.0 0.0.0.0 172.30.3.1
!
no ip http server
!
voice-port 0/0/0:15
!
!
!
!
!
dial-peer voice 20 pots
description ALL FROM VoIP GO THERE
preference 5
service session
destination-pattern 32....
translate-outgoing calling 482
direct-inward-dial
port 0/0/0:15
forward-digits all
!
dial-peer voice 5 voip
description ALL FROM POTS GO THERE
huntstop
preference 1
destination-pattern 6..
translate-outgoing called 60
voice-class codec 2
session protocol sipv2
session target ipv4:172.30.3.118:5069
session transport udp
dtmf-relay rtp-nte
no vad
!
sip-ua
no redirection
retry invite 3
retry response 3
retry bye 3
retry cancel 3
retry register 3
timers trying 1000
timers expires 300000
timers connect 1000
timers disconnect 1000
timers prack 1000
timers comet 1000
timers rel1xx 1000
timers refer 1000
timers info 1000
timers register 150
sip-server ipv4:172.30.3.118:5069
!

Debug Asterisk:

<------------>
    -- Executing [482600@from_cisco:1] Macro("SIP/172.30.3.119-081dedc0", "channel_sipnet|SIP/pri1/600") in new stack
    -- Executing [s@macro-channel_sipnet:1] NoOp("SIP/172.30.3.119-081dedc0", "Call To 482600 on this path SIP/pri1/600") in new stack
    -- Executing [s@macro-channel_sipnet:2] Dial("SIP/172.30.3.119-081dedc0", "SIP/pri1/600|20|m(wav)") in new stack
    -- Called pri1/600
Audio is at 172.30.3.118 port 6164
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.30.3.119:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP  172.30.3.119:5060;branch=z9hG4bK6EB113D;received=172.30.3.119
From: <sip:172.30.3.119>;tag=EC5D6E8-268E
To: <sip:482600@172.30.3.118>;tag=as3561ebe4
Call-ID: FC516BC4-D48411DD-A592EA68-5A27931F@172.30.3.119
CSeq: 101 INVITE
User-Agent: SakhaTelecom
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:482600@172.30.3.118:5069>
Content-Type: application/sdp
Content-Length: 239

v=0
o=root 29711 29711 IN IP4 172.30.3.118
s=session
c=IN IP4 172.30.3.118
t=0 0
m=audio 6164 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

<------------>
    -- Started music on hold, class 'wav', on SIP/172.30.3.119-081dedc0
    -- SIP/pri1-081e0350 is ringing
Sending to 172.30.3.119 : 5060 (no NAT)

<--- Reliably Transmitting (no NAT) to 172.30.3.119:5060 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP  172.30.3.119:5060;branch=z9hG4bK6EB113D;received=172.30.3.119
From: <sip:172.30.3.119>;tag=EC5D6E8-268E
To: <sip:482600@172.30.3.118>;tag=as3561ebe4
Call-ID: FC516BC4-D48411DD-A592EA68-5A27931F@172.30.3.119
CSeq: 101 INVITE
User-Agent: SakhaTelecom
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0

<------------>

<--- Transmitting (no NAT) to 172.30.3.119:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP  172.30.3.119:5060;branch=z9hG4bK6EB113D;received=172.30.3.119
From: <sip:172.30.3.119>;tag=EC5D6E8-268E
To: <sip:482600@172.30.3.118>;tag=as3561ebe4
Call-ID: FC516BC4-D48411DD-A592EA68-5A27931F@172.30.3.119
CSeq: 101 CANCEL
User-Agent: SakhaTelecom
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:482600@172.30.3.118:5069>
Content-Length: 0

<------------>
    -- Stopped music on hold on SIP/172.30.3.119-081dedc0
  == Spawn extension (macro-channel_sipnet, s, 2) exited non-zero on 'SIP/172.30.3.119-081dedc0' in macro 'channel_sipnet'
  == Spawn extension (macro-channel_sipnet, s, 2) exited non-zero on 'SIP/172.30.3.119-081dedc0'

<--- SIP read from 172.30.3.119:5060 --->
ACK sip:482600@172.30.3.118:5069 SIP/2.0
Via: SIP/2.0/UDP  172.30.3.119:5060;branch=z9hG4bK6EB113D
From: <sip:172.30.3.119>;tag=EC5D6E8-268E
To: <sip:482600@172.30.3.118>;tag=as3561ebe4
Date: Mon, 29 Dec 2008 02:12:32 GMT
Call-ID: FC516BC4-D48411DD-A592EA68-5A27931F@172.30.3.119
Max-Forwards: 1
CSeq: 101 ACK
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog 'FC516BC4-D48411DD-A592EA68-5A27931F@172.30.3.119' Method: ACK
[Dec 29 11:08:58] WARNING[29758]: chan_sip.c:12949 handle_response: Remote host can't match request CANCEL to call '0dcc145806495fb56ee4d0a46abce259@172.30.3.118'. Giving up.
Reliably Transmitting (no NAT) to 172.30.3.119:5060:
OPTIONS sip:172.30.3.119 SIP/2.0
Via: SIP/2.0/UDP 172.30.3.118:5069;branch=z9hG4bK5592b282;rport
From: "asterisk" <sip:asterisk@172.30.3.118:5069>;tag=as2062e0c8
To: <sip:172.30.3.119>
Contact: <sip:asterisk@172.30.3.118:5069>
Call-ID: 606d4b6c44edcd35654e06fd60d59404@172.30.3.118
CSeq: 102 OPTIONS
User-Agent: SakhaTelecom
Max-Forwards: 70
Date: Mon, 29 Dec 2008 02:09:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0

---
wsh*CLI> sip set debug peer
<--- SIP read from 172.30.3.119:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.3.118:5069;branch=z9hG4bK5592b282;rport
From: "asterisk" <sip:asterisk@172.30.3.118:5069>;tag=as2062e0c8
To: <sip:172.30.3.119>;tag=EC6278C-17D4
Date: Mon, 29 Dec 2008 02:12:53 GMT
Call-ID: 606d4b6c44edcd35654e06fd60d59404@172.30.3.118
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 OPTIONS
Supported: 100rel,replaces
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER
Accept: application/sdp
Allow-Events: telephone-event
Content-Length: 167
Content-Type: application/sdp

v=0
o=CiscoSystemsSIP-GW-UserAgent 9260 6822 IN IP4 172.30.3.119
s=SIP Call
c=IN IP4 172.30.3.119
t=0 0
m=audio 0 RTP/AVP 18 0 8 4 2 15 3
c=IN IP4 172.30.3.119

<------------->
--- (14 headers 7 lines) ---
Really destroying SIP dialog '606d4b6c44edcd35654e06fd60d59404@172.30.3.118' Method: OPTIONS