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.
Вариант для распечатки
Маршрутизаторы CISCO и др. оборудование. (Public)
(??) on
26-Авг-05, 14:33 (MSK)
