Здравствуйте.

На ATA выставил UseSIP в 1, прописал GkOrProxy и UID0.

На 1760:
dial-peer voice 101 voip
destination-pattern 7051
voice-class codec 1
session protocol sipv2
session target ipv4:10.190.0.30
dtmf-relay rtp-nte
no vad

Если на том конце ответили, то когда с АТА BYE проходит нормально. ccsip messages:

*Mar 1 09:00:47.606: Received:
BYE sip:989178424429@192.168.220.53 SIP/2.0
Via: SIP/2.0/UDP 10.190.0.30:5060
From: <sip:7051@192.168.220.53;user=phone>;tag=4243339994
To: <sip:989178424429@192.168.220.53;user=phone>;tag=1EECD3B-1FDE
Call-ID: 1541872116@10.190.0.30
CSeq: 2 BYE
User-Agent: Cisco ATA v2.15 ata186 (020723a)
Content-Length: 0

*Mar 1 09:00:47.618: Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.190.0.30:5060
From: <sip:7051@192.168.220.53;user=phone>;tag=4243339994
To: <sip:989178424429@192.168.220.53;user=phone>;tag=1EECD3B-1FDE
Date: Mon, 01 Mar 1993 09:00:47 GMT
Call-ID: 1541872116@10.190.0.30
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
CSeq: 2 BYE

А вот если там трубку не взяли, то телефон продолжает звонить

*Mar 1 09:04:06.500: Received:
CANCEL sip:989178424429@192.168.220.53;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.190.0.30:5060
From: <sip:7051@192.168.220.53;user=phone>;tag=3455357884
To: <sip:989178424429@192.168.220.53;user=phone>;tag=1F1FA10-B03
Call-ID: 278137398@10.190.0.30
CSeq: 1 CANCEL
User-Agent: Cisco ATA v2.15 ata186 (020723a)
Content-Length: 0

*Mar 1 09:04:06.504: Sent:
SIP/2.0 481 Call Leg/Transaction Does Not Exist
Via: SIP/2.0/UDP 10.190.0.30:5060
From: <sip:7051@192.168.220.53;user=phone>;tag=3455357884
To: <sip:989178424429@192.168.220.53;user=phone>;tag=1F1FA10-B03
Call-ID: 278137398@10.190.0.30
CSeq: 1 CANCEL
Content-Length: 0

Скажите, пожалуйста, куда копать. Поиском по cisco.com ничего полезного не нашел...