СвязьПроект

Добрый день. Шлюз настроил звонки проходят во все стороны. на всех FXO портах callerid enable, пробовал все type callerid. Результат один, шлюз не посылает в Астериск callerid. Как исправить данную ситуацию?

Учимся пользоваться поиском
http://www.xdsl.ru/svpro/viewtopic.php?t=414

все прям такие умные.... Видел я эту тему, в дебаге номера у меня нет. А вот что пишет:

caller-id type bellcore:
AP1005(config)# [90709.235] VM(0/3/0) Rx FXO Ring Actv
[90709.235] VM(0/3/0) Tx RING_IND
[90709.995] VM(0/3/0) Rx FXO Ring Idle
[90709.995] VM(0/3/0) CID enable
[90709.995] VM(0/3/0) vopp enable
[90709.995] VM(0/3/0) play mute
[90711.795] VM(0/3/0) Rx FXO Ring Actv
[90711.795] VM(0/3/0) vopp idle
[90711.795] VM(0/3/0) FXO OnHook
[90711.795] VM(0/3/0) Tx OFFHOOK_IND
[90711.795] VM(0/3/0) set T38 mode STD
[90711.795] VM(0/3/0) Fax rate 9600
155 <SIP 0> : No authentication information available
156 <SIP 380> : Send INVITE Request
[90711.850] RTA(0/3/0) Rx RS_LISTEN_REQ callId=380 ssId=1 G711U
peer=0.0.0.0 mp=23026/23027 hp=0/0
157 <SIP 380> : Receive 100 Trying
158 <SIP 380> : Transaction (84 INVITE) proceeding
159 <SIP 380> : Receive 180 Ringing
160 <SIP 380> : Transaction (84 INVITE) proceeding
[90711.925] RTA(0/3/0) Rx CC_ALERT_RSP peerId(0/0/0)
[90712.555] VM(0/3/0) Rx FXO Ring Idle
161 <SIP 380> : Receive 200 OK
[90713.765] RTA(0/3/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0
VAD_CTRL 0
[90713.765] VM(0/3/0) VAD disable
[90713.770] RTA(0/3/0) Rx CC_CONNECT_RSP peerId(0/0/0)
[90713.770] VM(0/3/0) FXO OffHook
[90713.770] VM(0/3/0) vopp enable
[90713.770] VM(0/3/0) Fax enable
162 <SIP 380> : Get SIP Audio MediaFormat : 0
[90713.770] RTA(0/3/0) Rx RS_OPEN_REQ callId=380 ssId=1 G711U
peer=192.168.0.7 mp=23026/23027 hp=19280/19281
[90713.775] VM(0/3/0) codec same G711U
[90713.775] RTA(0/3/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0
DTMF_CTRL 0
[90713.775] VM(0/3/0) DTMF disable
163 <SIP 380> : Received INVITE OK response
164 <SIP 380> : Send ACK Request
165 <SIP 380> : Check Event Relation
166 <SIP 380> : Set Terminated Success for 84 INVITE
167 <SIP 380> : Receive BYE Request
168 <SIP 380> : ReleaseWithNothing
[90718.455] RTA(0/3/0) Rx RS_CLOSE_REQ callId=380 ssId=1 dir=reve
[90718.455] RTA(0/3/0) Rx RS_CLOSE_REQ callId=380 ssId=1 dir=forw
[90718.455] RTA(0/3/0) close Media socket
[90718.455] RTA(0/3/0) close RTCP socket
[90718.455] RTA(0/3/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0
DTMF_STOP
[90718.455] VM(0/3/0) play mute
[90718.455] RTA(0/3/0) Rx CC_DISCONN_REQ CZ=0, peerId(0/0/0)
[90718.460] VM(0/3/0) vopp idle
[90718.460] VM(0/3/0) FXO OnHook
[90718.460] VM(0/3/0) Tx DISCONN_CNF


caller-id type etsi:
AP1005(config-voice-port-0/3)# [91051.365] VM(0/3/0) Rx FXO Ring Actv
[91051.365] VM(0/3/0) Tx RING_IND
[91052.125] VM(0/3/0) Rx FXO Ring Idle
[91052.125] VM(0/3/0) CID enable
[91052.125] VM(0/3/0) vopp enable
[91052.125] VM(0/3/0) play mute
[91053.325] VM(0/3/0) Rx CID DATA [L=40] 80 01 39 02 01 05 08 06 30 08 34 08 31 08 37 08 31 07 34 07 34 07 38 07 02 05 0b 06 38 09 39 09 31 09 38 09 35 09 36 09
[91053.490] VM(0/3/0) Rx CID DATA [L=40] 30 09 39 09 30 09 34 09 37 09 03 05 07 06 32 10 32 10 30 10 39 10 32 10 31 10 33 10 11 05 01 06 01 12 16 05 01 06 0a 16
[91053.655] VM(0/3/0) Rx CID DATA [L=42] 30 05 11 06 45 0e 57 0e 53 0e 44 0e 20 0e 66 0e 72 0e 6f 0e 6d 0e 20 0e 53 0e 49 0e 45 0e 4d 0e 45 0e 4e 0e 53 0e 2e 03 00 0f
[91053.925] VM(0/3/0) Rx FXO Ring Actv
[91053.925] VM(0/3/0) vopp idle
[91053.925] VM(0/3/0) FXO OnHook
[91053.925] VM(0/3/0) Rx CID [L=122] 80 01 39 02 01 05 08 06 30 08 34 08 31 08 37 08 31 07 34 07 34 07 38 07 02 05 0b 06 38 09 39 09 31 09 38 09 35 09 36 09 30 09 39 09 30 09 34 09 37 09 03 05 07 06 32 10 32 10 30 10 39 10 32 10 31 10 33 10 11 05 01 06 01 12 16 05 01 06 0a 16 30 05 11 06 45 0e 57 0e 53 0e 44 0e 20 0e 66 0e 72 0e 6f 0e 6d 0e 20 0e 53 0e 49 0e 45 0e 4d 0e 45 0e 4e 0e 53 0e 2e 03 00 0f
[91053.940] VM(0/3/0) Tx CID_DATA_IND [L=59] 80 39 01 08 30 34 31 37 31 34 34 38 02 0b 38 39 31 38 35 36 30 39 30 34 37 03 07 32 32 30 39 32 31 33 11 01 01 16 01 0a 30 11 45 57 53 44 20 66 72 6f 6d 20 53 49 45 4d 45 4e 53
[91053.940] VM(0/3/0) Tx OFFHOOK_IND
[91053.945] VM(0/3/0) set T38 mode STD
[91053.945] VM(0/3/0) Fax rate 9600
170 <SIP 0> : No authentication information available
171 <SIP 381> : Send INVITE Request
[91054.000] RTA(0/3/0) Rx RS_LISTEN_REQ callId=381 ssId=1 G711U
peer=0.0.0.0 mp=23028/23029 hp=0/0
172 <SIP 381> : Receive 100 Trying
173 <SIP 381> : Transaction (85 INVITE) proceeding
174 <SIP 381> : Receive 180 Ringing
175 <SIP 381> : Transaction (85 INVITE) proceeding
[91054.080] RTA(0/3/0) Rx CC_ALERT_RSP peerId(0/0/0)
[91054.645] VM(0/3/0) Rx FXO Ring Idle
176 <SIP 381> : Receive 200 OK
[91056.695] RTA(0/3/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0
VAD_CTRL 0
[91056.695] VM(0/3/0) VAD disable
[91056.695] RTA(0/3/0) Rx CC_CONNECT_RSP peerId(0/0/0)
[91056.695] VM(0/3/0) FXO OffHook
[91056.695] VM(0/3/0) vopp enable
[91056.695] VM(0/3/0) Fax enable
177 <SIP 381> : Get SIP Audio MediaFormat : 0
[91056.700] RTA(0/3/0) Rx RS_OPEN_REQ callId=381 ssId=1 G711U
peer=192.168.0.7 mp=23028/23029 hp=15922/15923
[91056.700] VM(0/3/0) codec same G711U
[91056.700] RTA(0/3/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0
DTMF_CTRL 0
[91056.700] VM(0/3/0) DTMF disable
178 <SIP 381> : Received INVITE OK response
179 <SIP 381> : Send ACK Request
180 <SIP 381> : Check Event Relation
181 <SIP 381> : Set Terminated Success for 85 INVITE
182 <SIP 381> : Receive BYE Request
183 <SIP 381> : ReleaseWithNothing
[91063.235] RTA(0/3/0) Rx RS_CLOSE_REQ callId=381 ssId=1 dir=reve
[91063.235] RTA(0/3/0) Rx RS_CLOSE_REQ callId=381 ssId=1 dir=forw
[91063.235] RTA(0/3/0) close Media socket
[91063.235] RTA(0/3/0) close RTCP socket
[91063.235] RTA(0/3/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0
DTMF_STOP
[91063.235] VM(0/3/0) play mute
[91063.235] RTA(0/3/0) Rx CC_DISCONN_REQ CZ=0, peerId(0/0/0)
[91063.235] VM(0/3/0) vopp idle
[91063.235] VM(0/3/0) FXO OnHook
[91063.235] VM(0/3/0) Tx DISCONN_CNF
184 <SIP 381> : Set Terminated Success for 102 BYE

А вот конфиг порта:

Voice port slot(0)/port(3)
line type = FXO
status = Idle
input gain = 0 db
output gain = 0 db
polarity inverse = disabled
number of rings to answer = 2
ring detect timeout = 7500 msec
ring detect timer = 300 msec
FXO hookflash out = 300 msec
clear down delay = 0 sec
clear down tone detect = enabled
tie connection = plar 100
description = Line_3
translate incoming called-number = -1
translate incoming calling-number = -1
comfort noise generation = disabled
dial tone generation = enabled
echo cancellation = enabled
announcement = enabled
low dtmf gain = -8
high dtmf gain = -5
caller ID = enabled
caller ID type = bellcore
caller ID NAME = disabled
DID Type = normal
busyout action = tone
backup busyout action = none
current callnumber = -1
holded callnumber = -1

конечно умные, но не телепаты, я не могу догадываться, что там у вас происходит.
И видно, что caller-id type etsi есть
Покажите ещё дебаг вместе с сипом:
deb voip call
deb rta ipc
deb voip sip

Разобрался, оказывается он пихает номер в CallerId(name), а не CallerId(num). Только вот, один определитель заработал, а на второй линии, от другого провайдера не работает, там в etsi ничего не высвечивается... (конфиг у второй линии такой же)

Да и еще проблема: Если говорить одновременно с собеседником, то перестаешь его слышать... Звук как-бы перебивается...

Прозвонил на станцию, поставил определитель в режим FSK. Все заработало!!!



Так никто не знает по поводу звука?

Может хотя бы конфигом порадуете?

А чего же не порадовать.

Using 3259 out of 65332 bytes
!
version 8.23F
!
hostname AP1005
!
dhcp-list 1 type server
dhcp-list 1 address server 10.1.1.2 10.1.1.126 255.255.255.128
!
!
no ip-share enable
ip-share interface net-side ether0.0
ip-share interface local-side ether1.0
!
interface ether0.0
ip address 192.168.0.8 255.255.255.0
description WAN
qos-control
line-ctrl full-duplex
!
interface ether1.0
no ip address
!
snmp name AP1005
!
no arp reset
!
route 0.0.0.0 0.0.0.0 192.168.0.1
!
logging event all
logging on
!
ntp refresh 5
ntp server 192.168.0.1
!
dnshost nameserver 192.168.0.1
!
!
!
!
!
! VoIP configuration.
!
!
! Voice service voip configuration.
!
voice service voip
fax protocol t38 redundancy 0
fax rate 9600
h323 call start fast
h323 call tunnel enable
voice-confirmed-connect 25
inband-ringback-tone
max-call 100
!
!
! Voice port configuration.
!
! FXO
voice-port 0/0
connection plar 100
description Line_0
ring number 100
ring detect-timeout 75
busyout action tone
caller-id enable
!
!
! FXO
voice-port 0/1
connection plar 100
description Line_1
ring number 100
ring detect-timeout 75
busyout action tone
caller-id enable
!
!
! FXO
voice-port 0/2
connection plar 100
description Line_2
ring number 100
ring detect-timeout 75
busyout action tone
caller-id enable
!
!
! FXO
voice-port 0/3
connection plar 100
description Line_3
ring number 100
ring detect-timeout 75
busyout action tone
caller-id enable
!
!
!
!
!
voice service voip
minimize-voip-port multiply 4
!
! Pots peer configuration.
!
dial-peer voice 1 pots
destination-pattern 1.T
port 0/0
forward-digits from 2
!
dial-peer voice 2 pots
destination-pattern 2.T
port 0/1
forward-digits from 2
!
dial-peer voice 3 pots
destination-pattern 3.T
port 0/2
forward-digits from 2
!
dial-peer voice 4 pots
destination-pattern 4.T
port 0/3
forward-digits from 2
!
dial-peer voice 100 pots
destination-pattern 0T
port 0/0
forward-digits from 1
!
dial-peer voice 110 pots
destination-pattern 5.T
port 0/1
forward-digits from 2
!
!
!
! Voip peer configuration.
!
dial-peer voice 1001 voip
destination-pattern 100
session target 192.168.0.7
session protocol sip
dtmf-relay rtp-2833
no vad
fax protocol t38 redundancy 0
!
!
!
!
!
!
! Gateway configuration.
!
gateway
h323-id voip.192.168.0.8
no ignore-msg-from-other-gk
!
!
! Codec classes configuration.
!
voice class codec 0
codec preference 1 g711alaw
codec preference 2 g729
!
!
!
! SIP UA configuration.
!
sip-ua
user-register
sip-username addpack
sip-password bingo
sip-server 192.168.0.7
!
!
! MGCP configuration.
!
mgcp
codec g711ulaw
!
!
! Tones
voice class clear-down-tone 0 430 0 745 740
!
voice class clear-down-tone 1 452 0 345 345
!
voice class dial-tone 430 0 10000 0 10000 0 -15
!
voice class ring-back-tone 430 0 1000 2000 1000 3000 -15
!
voice class line-busy-tone 425 0 350 350 0 0 -12
!
voice class reorder-tone 3000 0 10000 0 0 0 -15

И еще возник вопросик: Когда звонок, идет , например через первую линию, и номер занят (номер на который производится набор), то шлюз автоматически повторяет набор со второй линии. Надо это как-то отключить. Вот debug:

[681262.765] RTA(0/0/0) Rx RS_CLOSE_REQ callId=2065 ssId=1 dir=reve
[681262.765] RTA(0/0/0) Rx RS_CLOSE_REQ callId=2065 ssId=1 dir=forw
[681262.765] RTA(0/0/0) close Media socket
[681262.765] RTA(0/0/0) close RTCP socket
[681262.765] RTA(0/0/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0
DTMF_STOP
[681262.765] VM(0/0/0) play mute
[681262.765] RTA(0/0/0) Rx CC_DISCONN_REQ CZ=0, peerId(0/0/0)
[681262.765] VM(0/0/0) vopp idle
[681262.765] VM(0/0/0) FXO OnHook
[681262.765] VM(0/0/0) Tx DISCONN_CNF
14161 <SIP 2065> : Set Terminated Success for 103 BYE
14162 <SIP 2066> : Receive INVITE Request
[681329.125] RTA(0/0/0) Rx CC_OFFHOOK_REQ [32 34 33 36 35 36 35 ] peerId(-1)
[681329.125] VM(0/0/0) FXO OffHook
[681329.125] VM(0/0/0) vopp enable
[681329.125] VM(0/0/0) set T38 mode STD
[681329.125] VM(0/0/0) Fax rate 9600
14163 <SIP 2066> : SetAlerting
[681329.125] RTA(0/0/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0
VAD_CTRL 0
[681329.125] VM(0/0/0) VAD disable
14164 <SIP 2066> : Add Local Audio MediaFormat : 8
[681329.135] VM(0/0/0) Rx FXO Ring Actv
[681329.170] RTA(0/0/0) Rx RS_OPEN_REQ callId=2066 ssId=1 G711A
peer=192.168.0.7 mp=23002/23003 hp=19848/19849
[681329.170] VM(0/0/0) vopp idle
[681329.170] VM(0/0/0) start codec replace timer to G711A
[681329.235] VM(0/0/0) vopp enable
[681329.235] VM(0/0/0) codec replaced to G711A
[681329.385] VM(0/0/0) Rx FXO Ring Idle
[681330.125] VM(0/0/0) play digit '2'
[681330.275] VM(0/0/0) play mute
[681330.375] VM(0/0/0) play digit '4'
[681330.525] VM(0/0/0) play mute
[681330.625] VM(0/0/0) play digit '3'
[681330.775] VM(0/0/0) play mute
[681330.875] VM(0/0/0) play digit '6'
[681331.025] VM(0/0/0) play mute
[681331.125] VM(0/0/0) play digit '5'
[681331.275] VM(0/0/0) play mute
[681331.375] VM(0/0/0) play digit '6'
[681331.525] VM(0/0/0) play mute
[681331.625] VM(0/0/0) play digit '5'
[681331.775] VM(0/0/0) play mute
[681331.875] VM(0/0/0) Fax enable
[681331.875] VM(0/0/0) play mute
[681333.755] VM(0/0/0) listen LineBusy tone
[681333.755] VM(0/0/0) vopp idle
[681333.760] VM(0/0/0) FXO OnHook
[681333.760] VM(0/0/0) Tx DISCONN_CNF
[681333.770] RTA(0/0/0) Rx RS_CLOSE_REQ callId=2066 ssId=1 dir=forw
[681333.770] RTA(0/0/0) close Media socket
[681333.770] RTA(0/0/0) close RTCP socket
14165 <SIP 2066> : Receive ACK Request
14166 <SIP 2067> : Receive INVITE Request
14167 <CEP 000100> : InitiateOutCall : calledNum(2436565), callingNum(206), callerPort(ffffffff) type(FXO)
[681333.925] RTA(0/1/0) Rx CC_OFFHOOK_REQ [32 34 33 36 35 36 35 ] peerId(-1)
[681333.925] VM(0/1/0) FXO OffHook
[681333.925] VM(0/1/0) vopp enable
14168 <CEP 000100> : Outbound call to CEP callId(00000000-0000-0000-0000-000000000000) callNum(2067)
[681333.925] VM(0/1/0) set T38 mode STD
[681333.925] VM(0/1/0) Fax rate 9600
14169 <SIP 2067> : SetAlerting
14170 <Call 2067> : PreConnected from(100)
[681333.930] RTA(0/1/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0
VAD_CTRL 0
[681333.930] VM(0/1/0) VAD disable
14171 <SIP 2067> : Add Local Audio MediaFormat : 8
[681333.935] VM(0/1/0) Rx FXO Ring Actv
[681333.970] RTA(0/1/0) Rx RS_OPEN_REQ callId=2067 ssId=1 G711A
peer=192.168.0.7 mp=23004/23005 hp=14786/14787
[681333.975] VM(0/1/0) vopp idle
[681333.975] VM(0/1/0) start codec replace timer to G711A
[681334.035] VM(0/1/0) vopp enable
[681334.035] VM(0/1/0) codec replaced to G711A
[681334.195] VM(0/1/0) Rx FXO Ring Idle
[681334.925] VM(0/1/0) play digit '2'
[681335.075] VM(0/1/0) play mute
[681335.175] VM(0/1/0) play digit '4'
[681335.325] VM(0/1/0) play mute
[681335.425] VM(0/1/0) play digit '3'
[681335.575] VM(0/1/0) play mute
[681335.675] VM(0/1/0) play digit '6'
[681335.825] VM(0/1/0) play mute
[681335.925] VM(0/1/0) play digit '5'
[681336.075] VM(0/1/0) play mute
[681336.175] VM(0/1/0) play digit '6'
[681336.325] VM(0/1/0) play mute
[681336.425] VM(0/1/0) play digit '5'
[681336.575] VM(0/1/0) play mute
[681336.675] VM(0/1/0) Fax enable
[681336.675] VM(0/1/0) play mute
14172 <SIP 2066> : Set Terminated Success for 102 INVITE
14173 <SIP 2067> : Receive CANCEL Request
14174 <SIP 2067> : Cancel received
[681339.460] RTA(0/1/0) Rx RS_CLOSE_REQ callId=2067 ssId=1 dir=forw
[681339.460] RTA(0/1/0) close Media socket
[681339.460] RTA(0/1/0) close RTCP socket
14175 <Call 2067> : Terminated from(fffffffe) this(Remote:CallClear) before(NULL) forced(0)
14176 <CEP 000100> : DisconnectCall at Busy
14177 <CEP 000100> : StopSignal
[681339.460] RTA(0/1/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0
DTMF_STOP
[681339.460] VM(0/1/0) play mute
14178 <CEP 000100> : Disconnect (0)
[681339.465] RTA(0/1/0) Rx CC_DISCONN_REQ CZ=0, peerId(0/0/0)
[681339.465] VM(0/1/0) vopp idle
[681339.465] VM(0/1/0) FXO OnHook
[681339.465] VM(0/1/0) Tx DISCONN_CNF
14179 <SIP 2067> : Receive ACK Request
14180 <CEP 000100> : Disconnected(16) at Disconnecting
14181 <SIP 2067> : Set Terminated Success for 102 INVITE

Добавьте в потс пиры команду huntstop