Следующая ситуация: есть cisco 2610xm 12.2(11)T10 - это шлюз между ISDN и voip ,также есть несколько ата186 Для начала я хотел бы чтоб проходили  звонки с одной АТА на другую через маршрутизатор. С текущей конфигурацией звонок с АТА на АТА отбивается. Вот что у меня на cisco:
call rsvp-sync
!
!
mgcp profile default
!
dial-peer cor custom
!
!
dial-peer voice 5000 voip
destination-pattern 5000
session target ipv4:x.x.x.1 - ip ata1
!
dial-peer voice 5001 voip
destination-pattern 5001
session target ipv4:x.x.x.2 - ip ata2  

на аташках :
static ip:x.x.x.1
netmask:255.255.255.0
staticroute:ip cisco
gateway:ip cisco
uid0:5000

на cisco : sh dial-peer voice 5000
VoiceOverIpPeer5000
        information type = voice,
        description = `',
        tag = 5000, destination-pattern = `5000',
        answer-address = `', preference=0,
        CLID Restriction = None
        CLID Network Number = `'
        CLID Second Number sent
        source carrier-id = `', target carrier-id = `',
        source trunk-group-label = `',  target trunk-group-label = `',
        numbering Type = `unknown'
        group = 5000, Admin state is up, Operation state is up,
        incoming called-number = `', connections/maximum = 0/unlimited,
        DTMF Relay = disabled,
        modem transport = system,
        huntstop = disabled,
        in bound application associated: 'DEFAULT'
        out bound application associated: ''
        dnis-map =
        permission :both
        incoming COR list:maximum capability
        outgoing COR list:minimum requirement
        Translation profile (Incoming):
        Translation profile (Outgoing):
        incoming call blocking:
        translation-profile = `'
        disconnect-cause = `no-service'
        type = voip, session-target = `ipv4:10.12.1.200',
        technology prefix:
        settle-call = disabled
        ip media DSCP = ef, ip signaling DSCP = af31, UDP checksum = disabled,
        session-protocol = cisco, session-transport = system, req-qos = best-eff
ort,
        acc-qos = best-effort,
        RTP dynamic payload type values: NTE = 101
        Cisco: NSE=100, fax=96, fax-ack=97, dtmf=121, fax-relay=122
               CAS=123, ClearChan=125, PCM switch over u-law=0,A-law=8
        fax rate = voice,   payload size =  20 bytes
        fax protocol = system
        fax-relay ecm enable
        fax NSF = 0xAD0051 (default)
        codec = g729r8,   payload size =  20 bytes,
        Expect factor = 0, Icpif = 20,
        Playout Mode is set to adaptive,
        Initial 60 ms, Max 300 ms
        Playout-delay Minimum mode is set to default, value 40 ms
        Max Redirects = 1, signaling-type = cas,
        VAD = enabled, Poor QOV Trap = disabled,
        Source Interface = NONE
        voice class sip url = system,
        voice class sip rel1xx = system,
        voice class perm tag = `'
        Time elapsed since last clearing of voice call statistics never
        Connect Time = 0, Charged Units = 0,
        Successful Calls = 0, Failed Calls = 0, Incomplete Calls = 0
        Accepted Calls = 0, Refused Calls = 4,
        Last Disconnect Cause is "3   ",
        Last Disconnect Text is "no route to destination (3)",
        Last Setup Time = 136874355.
И еще если на телефоне подключенный к аташке просто поднять трубку , то примерно через 10 сек  начинаются короткие гудки . Gatekeeper'a в сети нет.Аташки находятся в одной сети.

Если кто-то сталкивался с подобной ситуацией, то напишите в чем дело.