Здравствуйте,есть cisco 28xx,cme

Настроен Sip

Линия зарегистрирована,входящие проходят.

Line                              peer        expires(sec)  registered
================================  ==========  ============  ==========
4956608185-1                      -1          1224          yes

sip-ua
credentials username 4956608185-1 password 7 gggggggggggggggggggggggggggg realm asterisk
retry invite 2
retry response 2
retry bye 2
retry cancel 2
registrar ipv4:87.255.9.250 expires 3600
sip-server ipv4:87.255.9.250

dial-peer voice 999 voip
description ###ALL###
translation-profile outgoing SIP_OUT
destination-pattern 9.T
session protocol sipv2
session target sip-server
codec g711ulaw
no vad

voice translation-rule 1
rule 1 /302/ /4956608185/
!
voice translation-rule 2
rule 1 /^9/ //
!
!
!
voice translation-profile SIP_IN
translate called 20
!
voice translation-profile SIP_OUT
translate calling 1
translate called 2

При наборе номера, к примеру 98......, отбой,лог sip такой:
Подскажите, пожалуйста,где косяк.....

*Nov 26 22:45:16.343: //662/BF16CA8E82EC/SIP/Media/sipSPICopyPeerDataToCCB: Firewall traversal is not enabled
*Nov 26 22:45:16.343: //662/BF16CA8E82EC/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 10.110.200.137
*Nov 26 22:45:16.343: //-1/xxxxxxxxxxxx/SIP/Media/sipSPIReserveRtpPort: reserved port 19174 for stream 1
*Nov 26 22:45:16.343: //662/BF16CA8E82EC/SIP/Media/sipSPIProcessRtpSessions: sipSPIProcessRtpSessions
*Nov 26 22:45:16.343: //662/BF16CA8E82EC/SIP/Media/sipSPIProcessRtpSessions: No active streams.
*Nov 26 22:45:16.347: //662/BF16CA8E82EC/SIP/Media/sipSPIProcessRtpSessions: sipSPIProcessRtpSessions
*Nov 26 22:45:16.347: //662/BF16CA8E82EC/SIP/Media/sipSPIAddStream: Adding stream 1 of type voice+dtmf (callid 662) to the VOIP RTP library
*Nov 26 22:45:16.347: //662/BF16CA8E82EC/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 10.110.200.137
*Nov 26 22:45:16.347: //662/BF16CA8E82EC/SIP/Media/sipSPIUpdateRtcpSession: sipSPIUpdateRtcpSession for m-line 1
*Nov 26 22:45:16.347: //662/BF16CA8E82EC/SIP/Media/sipSPIUpdateRtcpSession: rtcp_session info
        laddr = 10.110.200.137, lport = 19174, raddr = 0.0.0.0, rport=0, do_rtcp=FALSE
        src_callid = 662, dest_callid = -1, stream type = voice+dtmf, stream direction = RECVONLY
        media_ip_addr =  - , vrf tableid = 0 media_addr_type = 1
*Nov 26 22:45:16.347: //662/BF16CA8E82EC/SIP/Media/sipSPIUpdateRtcpSession: No rtp session, creating a new one
*Nov 26 22:45:16.347: //662/BF16CA8E82EC/SIP/Media/sipSPICreateRtpSession: stun is disabled
*Nov 26 22:45:16.347: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:89153457820@87.255.9.250:5060 SIP/2.0
Date: Mon, 26 Nov 2012 22:45:16 GMT
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
From: "Davydov N.V." <sip:4956608185@87.255.9.250>;tag=43551C-193A
Allow-Events: telephone-event
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE:  1800
Remote-Party-ID: "Davydov N.V." <sip:4956608185@10.110.200.137>;party=calling;screen=no;privacy=off
Cisco-Guid: 3205941902-928059874-2196551658-898321195
Timestamp: 1353969916
Content-Length: 197
User-Agent: Cisco-SIPGateway/IOS-12.x
To: <sip:89153457820@87.255.9.250>
Contact: <sip:4956608185@10.110.200.137:5060>
Expires: 180
Content-Disposition: session;handling=required
Content-Type: application/sdp
Call-ID: C81816A7-375111E2-82F1B7EA-358B4B2B@10.110.200.137
Via: SIP/2.0/UDP 10.110.200.137:5060;branch=z9hG4bK2B1A2A
CSeq: 101 INVITE
Max-Forwards: 70

v=0
o=CiscoSystemsSIP-GW-UserAgent 3592 9197 IN IP4 10.110.200.137
s=SIP Call
c=IN IP4 10.110.200.137
t=0 0
m=audio 19174 RTP/AVP 0
c=IN IP4 10.110.200.137
a=rtpmap:0 PCMU/8000
a=ptime:20

*Nov 26 22:45:16.351: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.110.200.137:5060;branch=z9hG4bK2B1A2A;received=10.110.200.137;rport=54831
From: "Davydov N.V." <sip:4956608185@87.255.9.250>;tag=43551C-193A
To: <sip:89153457820@87.255.9.250>;tag=as34d279ec
Call-ID: C81816A7-375111E2-82F1B7EA-358B4B2B@10.110.200.137
CSeq: 101 INVITE
Server: Asterisk PBX 1.8.15.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7f37b9c0"
Content-Length: 0

*Nov 26 22:45:16.355: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:89153457820@87.255.9.250:5060 SIP/2.0
Date: Mon, 26 Nov 2012 22:45:16 GMT
From: "Davydov N.V." <sip:4956608185@87.255.9.250>;tag=43551C-193A
Allow-Events: telephone-event
Content-Length: 0
To: <sip:89153457820@87.255.9.250>;tag=as34d279ec
Call-ID: C81816A7-375111E2-82F1B7EA-358B4B2B@10.110.200.137
Via: SIP/2.0/UDP 10.110.200.137:5060;branch=z9hG4bK2B1A2A
CSeq: 101 ACK
Max-Forwards: 70