1 | изначальная версия редактировать | |
Всем добрый день.
2 | No.2 Revision редактировать |
Всем добрый день.
При входящем вызове не приходит caller id. Уточнил у провайдера (Ростелеком), что на этих номерах действительно включен Caller ID. Сказали что в настройках стоит DTMF. Подключал к линиям обычный Panasonic, номера действительно определяются. Но на Аддпаке не хотят. Вот результат звонка (спользовал команды debug voip call, debug rta ipc):
robrobros_FXO_masl# [5424.935] VM(0/3/0) Rx FXO Ring Actv
[5424.935] VM(0/3/0) Tx RING_IND
3261 <CEP 000300> : Call Received
[5425.065] VM(0/3/0) Rx FXO Ring Idle
[5425.065] VM(0/3/0) Rx FXO Ring Ignore
[5425.065] VM(0/3/0) Tx DISCONN_CNF
3262 <CEP 000300> : Disconnected(16) at Busy
[5425.075] VM(0/3/0) Rx FXO Ring Actv ignore
[5426.285] VM(0/3/0) Rx FXO Ring Actv
[5426.285] VM(0/3/0) Tx RING_IND
3263 <CEP 000300> : Call Received
[5427.395] VM(0/3/0) Rx FXO Ring Idle
[5427.395] VM(0/3/0) CID enable
[5427.395] VM(0/3/0) vopp enable
[5427.395] VM(0/3/0) play mute
[5427.405] VM(0/3/0) Rx FXO Ring Actv ignore
[5431.485] VM(0/3/0) Rx FXO Ring Actv
[5431.485] VM(0/3/0) vopp idle
[5431.485] VM(0/3/0) FXO OnHook
[5431.485] VM(0/3/0) Tx OFFHOOK_IND
3264 <CEP 000300> : Call Initiated : calledNumber() crv(0) total(0)
3265 <Call 7> : ****************** Call Created status(InitiatedByFXO) *******************
3266 <CEP 000300> : Decode CID :
**3267 <CEP 000300> : Calling number()**
3268 <CEP 000300> : Call id(0f96bc54-346d-25ff-8017-0002a4040ea8) callNum(7)
3269 <Call 7> : MatchAllProcess After Sorted
<0> id(600) dest(.T) prefer(0) selected(6)
3270 <Call 7> : Initiate callee with dial-peer(.T) status(CalleeDeterminedAll) id(0f96bc54-346d-25ff-8017-0002a4040ea8)
3271 <NetEP 7> : InitiateOutCall: calledNum(371435) callingNum() target(sip-server)
3272 <NetEP 7> : DoCall: calledAddr(sip:371435@192.168.5.190:5060) callingAddr()
[5431.490] VM(0/3/0) set T38 mode STD
[5431.490] VM(0/3/0) Fax rate 9600
3273 <SIP 0> : No authentication information available
3274 <SIP 7> : Send INVITE Request
[5431.540] RTA(0/3/0) Rx RS_LISTEN_REQ callId=7 ssId=1 G711U
peer=0.0.0.0 mp=23014/23015 hp=0/0
3275 <SIP 7> : Receive 401 Unauthorized
3276 <SIP 7> : Transaction (17 INVITE) completed
3277 <SIP 7> : Send ACK Request
3278 <SIP 0> : No opaque in authentication
3279 <SIP 0> : Adding authentication information
3280 <SIP 7> : Send INVITE Request
3281 <SIP 7> : Receive 100 Trying
3282 <SIP 7> : Transaction (18 INVITE) proceeding
3283 <SIP 7> : Receive 200 OK
3284 <SIP 7> : Get SIP Audio MediaFormat : 8
3285 <SIP 7> : SetRemoteSocketInfo : ip(192.168.5.190) port(19630)
[5431.675] RTA(0/3/0) Rx RS_OPEN_REQ callId=7 ssId=1 G711A
peer=192.168.5.190 mp=23014/23015 hp=19630/19631
[5431.675] RTA(0/3/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0
DTMF_CTRL 1
[5431.675] VM(0/3/0) DTMF_RTP_RFC2833 enable
[5431.680] VM(0/3/0) DTMF_RTP_RFC2833 TxPT=0x65, RxPT=0x65
3286 <Call 7> : Connected from(fffffffe)
[5431.680] RTA(0/3/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0
VAD_CTRL 0
[5431.680] VM(0/3/0) VAD disable
[5431.680] VM(0/3/0) SID enable by CCC
[5431.680] RTA(0/3/0) Rx CC_CONNECT_RSP peerId(0/0/0)
[5431.680] VM(0/3/0) FXO OffHook
[5431.680] VM(0/3/0) vopp enable
[5431.680] VM(0/3/0) Fax enable
[5431.680] VM(0/3/0) play mute
3287 <NetEP 7> : Call with sip:371435@192.168.5.190 established
3288 <SIP 7> : Received INVITE OK response
3289 <SIP 7> : Send ACK Request
3290 <SIP 7> : Check Event Relation
3291 <SIP 7> : Set Terminated Success for 18 INVITE
[5431.725] VM(0/3/0) codec same G711A
[5431.725] VM(0/3/0) Rx RTP replace codec to G711A
[5431.795] VM(0/3/0) Rx FXO Ring Idle
[5431.805] VM(0/3/0) Rx FXO Ring Actv ignore
3292 <SIP 6> : Set Terminated Success for 14 INVITE
Как видно из дампа, Calling number() - пусто. Пробовал все пять типов caller id (etsi, etsi-dtmf, estsi-dtmf-prior-ring, bellcore, ntt), результат аналогичный. Поле Calling number() постоянно пустое.
Вот конфиг самого Аддпака:
!
version 8.30W
!
hostname probrobros_FXO_masl
!
!
no bridge spanning-tree
!
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 192.168.5.191 255.255.255.0
line-ctrl full-duplex
!
interface ether1.0
no ip address
!
snmp name AP1100F
snmp enable-trap dn-register 300 forcely-block
!
no arp reset
!
route 0.0.0.0 0.0.0.0 192.168.5.254
!
utilization cpu
utilization ethernet
!
logging command
logging event all
logging on
!
service tftpd
!
telnet-access 10.0.1.27 255.255.255.0
telnet-access 10.0.1.2 255.255.255.0
telnet-access 10.0.1.237 255.255.255.0
telnet-access 192.168.27.101 255.255.255.0
telnet-access 192.168.27.100 255.255.255.0
telnet-access 192.168.27.110 255.255.255.0
telnet-access host 192.168.6.2
!
!
!
!
!
! 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 port configuration.
!
! FXO
voice-port 0/0
input gain 4
output gain 4
connection plar 317451
description Trunk_1
ring detect-timeout 70
ring detect-timer 900
no comfort-noise
high-dtmf-gain -7
caller-id enable
caller-id type etsi
caller-id name disable
forced-clear-down -55 70
!
!
! FXO
voice-port 0/1
input gain 4
output gain 4
connection plar 321833
description Trunk_2
ring detect-timeout 40
ring detect-timer 900
no comfort-noise
high-dtmf-gain -7
caller-id enable
caller-id type etsi
caller-id name disable
forced-clear-down -55 70
!
!
! FXO
voice-port 0/2
input gain 4
output gain 4
connection plar 371438
description Trunk_3
ring detect-timeout 70
ring detect-timer 900
no comfort-noise
high-dtmf-gain -7
caller-id enable
caller-id type etsi
caller-id name disable
forced-clear-down -55 70
!
!
! FXO
voice-port 0/3
connection plar 371435
description Trunk_4
ring detect-timeout 70
ring detect-timer 900
no comfort-noise
caller-id enable
caller-id type etsi
caller-id name disable
forced-clear-down -55 70
!
!
!
!
! Pots peer configuration.
!
dial-peer voice 0 pots
destination-pattern 1T
port 0/0
user-name trunkout1
user-password ul7019158
preference 1
!
dial-peer voice 1 pots
destination-pattern 2T
port 0/1
user-name trunkout2
user-password ul7019158
preference 2
!
dial-peer voice 2 pots
destination-pattern 3T
port 0/2
user-name trunkout3
user-password ul7019158
preference 3
!
dial-peer voice 3 pots
destination-pattern 4T
port 0/3
user-name trunkout4
user-password ul7019158
preference 4
!
!
!
! Voip peer configuration.
!
dial-peer voice 600 voip
destination-pattern .T
session target 192.168.5.190
session protocol sip
voice-class codec 1
no vad
dtmf-relay rtp-2833
!
!
!
!
!
!
! Gateway configuration.
!
gateway
h323-id probrobros_FXO_masl
no ignore-msg-from-other-gk
!
!
! Codec classes configuration.
!
voice class codec 1
codec preference 1 g711alaw
codec preference 2 g711ulaw
!
!
!
! SIP UA configuration.
!
sip-ua
user-register
sip-server 192.168.27.110
sip-server 192.168.5.190
remote-party-id
register e164
!
!
! MGCP configuration.
!
mgcp
dtmf-relay rtp-2833
codec g711alaw
vad
!
!
! Tones
voice class clear-down-tone 0 425 0 300 300
!
!
!
!
voip-interface ether0.0
!
Люди добрые, помогите пожалуйста.
Проект компании "АТС Дизайн"
Asterisk® и Digium® являются зарегистрированными торговыми марками компании
Digium, Inc., США.
IP АТС Asterisk распространяется под лицензией
GNU GPL.