Досталась система(*+addpack ap1005 +tda-100) без какой либо тех документации.
Не известно что и куда подключено. Осмотреть все "медные" подключения в ближайшем будущем не состовляет возможности.
Пока только слинковал * с ap1005. Делаю дозвон с софтфона в ТФОП. Звонок конечно не идет. Перво наперво конечно надо грешить на то что воткнуто в FXO порты. Пчто неизвестно идеть "медь" из AP1005 в панас tda100, либо же сразу в ТФОП.
Собсвенно и возникли вопросы:
1. Как узнать что и в какой порт заходит и куда уходит на ap1005?
2. Если никак, то косвенные доказательства подключения addpack к панасу, как-то через включения дебага?
Связь ap1005 и панасоника еще никак не настраивал по этому буду рад если приведете пример как их слинковать. Возможно потом сам смогу ответить на п2.
Ну и на последок гляньте о великие гуру может все таки, что-то не дочитал не донастроил, что бы звонить.
debug ap1005 при звонке на номер ТФОП 42193
PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER: debug ap1005
21491 <Call 6> : MatchedAll
21492 <Call 6> : MatchAllProcess After Sorted
<0> id(0) dest(T) prefer(1) selected(5)
<1> id(1) dest(T) prefer(2) selected(0)
<2> id(2) dest(T) prefer(3) selected(0)
21493 <Call 6> : Initiate callee with dial-peer(T) status(CalleeDeterminedAll) id(00000000-0000-0000-0000-00000
0000000)
21494 <CEP 000000> : InitiateOutCall : calledNum(42193), callingNum(200), callerPort(ffffffff) type(FXO)
21495 <CEP 000000> : Outbound call to CEP callId(00000000-0000-0000-0000-000000000000) callNum(6)
21496 <SIP 6> : SetAlerting
21497 <Call 6> : PreConnected from(0)
21498 <SIP 6> : Add Local Audio MediaFormat : 8
21499 <CEP 000000> : Disconnected(16) at Busy
21500 <Call 6> : Terminated from(0) this(Local:CallClear) before(NULL) forced(0)
21501 <NetEP 6> : Call TO <200> terminated reason(Local:CallClear)
21502 <CEP 000000> : DisconnectCall at Idle
21503 <SIP 6> : Receive ACK Request
21504 <Time 0> : SIP_TREGISTER timer timeout.
21505 <SIP 0> : No authentication information available
21506 <SIP 3378> : Send REGISTER Request
21507 <Time 0> : SIP_TREGISTER timer timeout.
21508 <SIP 0> : No authentication information available
21509 <SIP 3379> : Send REGISTER Request
21510 <SIP 3378> : Receive 401 Unauthorized
21511 <SIP 3378> : Transaction (3378 REGISTER) completed
21512 <SIP 0> : No opaque in authentication
21513 <SIP 0> : Adding authentication information
21514 <SIP 3380> : Send REGISTER Request
21515 <SIP 3379> : Receive 401 Unauthorized
21516 <SIP 3379> : Transaction (3379 REGISTER) completed
21517 <SIP 0> : No opaque in authentication
21518 <SIP 0> : Adding authentication information
21519 <SIP 3381> : Send REGISTER Request
21520 <SIP 3380> : Receive 401 Unauthorized
21521 <SIP 3380> : Transaction (3380 REGISTER) completed
21522 <SIP 3381> : Already done REGISTER for Authentication Required.
21523 <SIP 3381> : Send SNMP TRAP (false with code - 401) previous false
21524 <SIP 3381> : Receive 401 Unauthorized
21525 <SIP 3381> : Transaction (3381 REGISTER) completed
21526 <SIP 0> : No opaque in authentication
21527 <SIP 0> : Adding authentication information
21528 <SIP 3382> : Send REGISTER Request
21529 <SIP 6> : Set Terminated Success for 102 INVITE
21530 <SIP 3382> : Receive 403 Forbidden (Bad auth)
21492 <Call 6> : MatchAllProcess After Sorted
<0> id(0) dest(T) prefer(1) selected(5)
<1> id(1) dest(T) prefer(2) selected(0)
<2> id(2) dest(T) prefer(3) selected(0)
21493 <Call 6> : Initiate callee with dial-peer(T) status(CalleeDeterminedAll) id(00000000-0000-0000-0000-00000
0000000)
21494 <CEP 000000> : InitiateOutCall : calledNum(42193), callingNum(200), callerPort(ffffffff) type(FXO)
21495 <CEP 000000> : Outbound call to CEP callId(00000000-0000-0000-0000-000000000000) callNum(6)
21496 <SIP 6> : SetAlerting
21497 <Call 6> : PreConnected from(0)
21498 <SIP 6> : Add Local Audio MediaFormat : 8
21499 <CEP 000000> : Disconnected(16) at Busy
21500 <Call 6> : Terminated from(0) this(Local:CallClear) before(NULL) forced(0)
21501 <NetEP 6> : Call TO <200> terminated reason(Local:CallClear)
21502 <CEP 000000> : DisconnectCall at Idle
21503 <SIP 6> : Receive ACK Request
21504 <Time 0> : SIP_TREGISTER timer timeout.
21505 <SIP 0> : No authentication information available
21506 <SIP 3378> : Send REGISTER Request
21507 <Time 0> : SIP_TREGISTER timer timeout.
21508 <SIP 0> : No authentication information available
21509 <SIP 3379> : Send REGISTER Request
21510 <SIP 3378> : Receive 401 Unauthorized
21511 <SIP 3378> : Transaction (3378 REGISTER) completed
21512 <SIP 0> : No opaque in authentication
21513 <SIP 0> : Adding authentication information
21514 <SIP 3380> : Send REGISTER Request
21515 <SIP 3379> : Receive 401 Unauthorized
21516 <SIP 3379> : Transaction (3379 REGISTER) completed
21517 <SIP 0> : No opaque in authentication
21518 <SIP 0> : Adding authentication information
21519 <SIP 3381> : Send REGISTER Request
21520 <SIP 3380> : Receive 401 Unauthorized
21521 <SIP 3380> : Transaction (3380 REGISTER) completed
21522 <SIP 3381> : Already done REGISTER for Authentication Required.
21523 <SIP 3381> : Send SNMP TRAP (false with code - 401) previous false
21524 <SIP 3381> : Receive 401 Unauthorized
21525 <SIP 3381> : Transaction (3381 REGISTER) completed
21526 <SIP 0> : No opaque in authentication
21527 <SIP 0> : Adding authentication information
21528 <SIP 3382> : Send REGISTER Request
21529 <SIP 6> : Set Terminated Success for 102 INVITE
21530 <SIP 3382> : Receive 403 Forbidden (Bad auth)
PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER: asterisk debug
== Using SIP RTP CoS mark 5
-- Executing [42193@phones:1] Dial("SIP/200-0000000a", "SIP/addpac/42193,100") in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/addpac/42193
-- SIP/addpac-0000000b is making progress passing it to SIP/200-0000000a
-- Got SIP response 480 "Temporarily Unavailable" back from 192.168.0.2:5060
-- SIP/addpac-0000000b is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing [42193@phones:2] Hangup("SIP/200-0000000a", "") in new stack
== Spawn extension (phones, 42193, 2) exited non-zero on 'SIP/200-0000000a'
-- Executing [42193@phones:1] Dial("SIP/200-0000000a", "SIP/addpac/42193,100") in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/addpac/42193
-- SIP/addpac-0000000b is making progress passing it to SIP/200-0000000a
-- Got SIP response 480 "Temporarily Unavailable" back from 192.168.0.2:5060
-- SIP/addpac-0000000b is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing [42193@phones:2] Hangup("SIP/200-0000000a", "") in new stack
== Spawn extension (phones, 42193, 2) exited non-zero on 'SIP/200-0000000a'
PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER:
[general]
[200]
type=friend
context=phones
host=dynamic
secret=200200
[201]
type=friend
context=phones
host=dynamic
secret=201201
[1005]
type=friend
context=phones
host=192.168.0.2
secret=10051005
[utda]
type=friend
context=phones
host=192.168.0.2
secret=utda
[addpac]
type=friend
host=192.168.0.2
nat=no
dtmfmode=rfc2833
context=phones
disallow=all
[200]
type=friend
context=phones
host=dynamic
secret=200200
[201]
type=friend
context=phones
host=dynamic
secret=201201
[1005]
type=friend
context=phones
host=192.168.0.2
secret=10051005
[utda]
type=friend
context=phones
host=192.168.0.2
secret=utda
[addpac]
type=friend
host=192.168.0.2
nat=no
dtmfmode=rfc2833
context=phones
disallow=all
PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER:
[globals]
[general]
[default]
exten => s,1,Verbose(1|Unrouted call handler)
exten => s,n,Answer()
exten => s,n,Wait(1)
exten => s,n,Playback(tt-weasels)
exten => s,n,Hangup()
[incoming_calls]
[internal]
exten => 500,1,Verbose(1,Echo test application)
exten => 500,n,Echo()
exten => 500,n,Hangup()
[phones]
;exten => 205,1,Dial(SIP/205,60)
;exten => 205,n,Hangup()
exten => _20X,1,Dial(SIP/${EXTEN},60)
exten => _20X,n,Hangup()
include => internal
include => external_calls
[external_calls]
;exten => _20X,1,Dial(SIP/${EXTEN},60)
;exten => _20X,n,Hangup()
exten => _4XXXX,1,Dial(SIP/addpac/${EXTEN},100)
exten => _4XXXX,n,HangUp
[general]
[default]
exten => s,1,Verbose(1|Unrouted call handler)
exten => s,n,Answer()
exten => s,n,Wait(1)
exten => s,n,Playback(tt-weasels)
exten => s,n,Hangup()
[incoming_calls]
[internal]
exten => 500,1,Verbose(1,Echo test application)
exten => 500,n,Echo()
exten => 500,n,Hangup()
[phones]
;exten => 205,1,Dial(SIP/205,60)
;exten => 205,n,Hangup()
exten => _20X,1,Dial(SIP/${EXTEN},60)
exten => _20X,n,Hangup()
include => internal
include => external_calls
[external_calls]
;exten => _20X,1,Dial(SIP/${EXTEN},60)
;exten => _20X,n,Hangup()
exten => _4XXXX,1,Dial(SIP/addpac/${EXTEN},100)
exten => _4XXXX,n,HangUp
PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER: conf
Using 3022 out of 65332 bytes
!
version 8.30I
!
hostname AP1005
!
!
no bridge spanning-tree
!
!
no ip-share enable
!
interface ether0.0
ip address 192.168.0.2 255.255.255.0
!
interface ether1.0
no ip address
!
snmp contact
snmp location
snmp name AP1005
no snmp trap-authentication
!
no arp reset
!
!
!
!
!
! 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
announcement language english
!
!
! Voice port configuration.
!
! FXO
voice-port 0/0
connection plar 200
description line-1
ring detect-timeout 40
no comfort-noise
busyout action tone
caller-id enable
!
!
! FXO
voice-port 0/1
connection plar 200
description line-2
ring detect-timeout 40
busyout action tone
caller-id enable
caller-id type etsi
!
!
! FXO
voice-port 0/2
connection plar 200
description line-3
ring detect-timeout 40
busyout action tone
caller-id enable
caller-id type etsi
!
!
! FXO
voice-port 0/3
connection plar 199
ring detect-timeout 100
busyout action tone
caller-id enable
caller-id type etsi
!
!
!
!
! Pots peer configuration.
!
dial-peer voice 0 pots
destination-pattern T
port 0/0
preference 1
huntstop
!
dial-peer voice 1 pots
destination-pattern T
port 0/1
preference 2
huntstop
!
dial-peer voice 2 pots
destination-pattern T
port 0/2
preference 3
huntstop
!
dial-peer voice 3 pots
destination-pattern 100
port 0/2
!
dial-peer voice 4 pots
destination-pattern 100
port 0/3
!
!
!
! Voip peer configuration.
!
dial-peer voice 5 voip
destination-pattern T
session target 192.168.0.32
session protocol sip
voice-class codec 0
dtmf-relay rtp-2833
no vad
!
dial-peer voice 1000 voip
destination-pattern .T
session target 192.168.0.3
session protocol sip
codec g711alaw
voice-class codec 1
dtmf-relay rtp-2833
no vad
huntstop
!
dial-peer voice 3001 voip
destination-pattern T
session target 192.168.0.3
session protocol sip
dtmf-relay rtp-2833
no vad
!
!
!
!
!
!
! Gateway configuration.
!
gateway
h323-id voip.192.168.0.2
no ignore-msg-from-other-gk
!
!
! Codec classes configuration.
!
voice class codec 0
codec preference 1 g711alaw
codec preference 2 g711ulaw
codec preference 3 g729
codec preference 4 g7231r53
codec preference 5 g7231r63
!
voice class codec 1
codec preference 1 g729
codec preference 2 g7231r63
codec preference 3 g7231r53
!
!
!
! SIP UA configuration.
!
sip-ua
user-register
sip-username utda
sip-password utda
sip-server 192.168.0.3
register e164
!
!
! MGCP configuration.
!
mgcp
codec g711ulaw
vad
!
!
! Tones
voice class clear-down-tone 0 425 0 355 355
!
!
!
!
!
version 8.30I
!
hostname AP1005
!
!
no bridge spanning-tree
!
!
no ip-share enable
!
interface ether0.0
ip address 192.168.0.2 255.255.255.0
!
interface ether1.0
no ip address
!
snmp contact
snmp location
snmp name AP1005
no snmp trap-authentication
!
no arp reset
!
!
!
!
!
! 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
announcement language english
!
!
! Voice port configuration.
!
! FXO
voice-port 0/0
connection plar 200
description line-1
ring detect-timeout 40
no comfort-noise
busyout action tone
caller-id enable
!
!
! FXO
voice-port 0/1
connection plar 200
description line-2
ring detect-timeout 40
busyout action tone
caller-id enable
caller-id type etsi
!
!
! FXO
voice-port 0/2
connection plar 200
description line-3
ring detect-timeout 40
busyout action tone
caller-id enable
caller-id type etsi
!
!
! FXO
voice-port 0/3
connection plar 199
ring detect-timeout 100
busyout action tone
caller-id enable
caller-id type etsi
!
!
!
!
! Pots peer configuration.
!
dial-peer voice 0 pots
destination-pattern T
port 0/0
preference 1
huntstop
!
dial-peer voice 1 pots
destination-pattern T
port 0/1
preference 2
huntstop
!
dial-peer voice 2 pots
destination-pattern T
port 0/2
preference 3
huntstop
!
dial-peer voice 3 pots
destination-pattern 100
port 0/2
!
dial-peer voice 4 pots
destination-pattern 100
port 0/3
!
!
!
! Voip peer configuration.
!
dial-peer voice 5 voip
destination-pattern T
session target 192.168.0.32
session protocol sip
voice-class codec 0
dtmf-relay rtp-2833
no vad
!
dial-peer voice 1000 voip
destination-pattern .T
session target 192.168.0.3
session protocol sip
codec g711alaw
voice-class codec 1
dtmf-relay rtp-2833
no vad
huntstop
!
dial-peer voice 3001 voip
destination-pattern T
session target 192.168.0.3
session protocol sip
dtmf-relay rtp-2833
no vad
!
!
!
!
!
!
! Gateway configuration.
!
gateway
h323-id voip.192.168.0.2
no ignore-msg-from-other-gk
!
!
! Codec classes configuration.
!
voice class codec 0
codec preference 1 g711alaw
codec preference 2 g711ulaw
codec preference 3 g729
codec preference 4 g7231r53
codec preference 5 g7231r63
!
voice class codec 1
codec preference 1 g729
codec preference 2 g7231r63
codec preference 3 g7231r53
!
!
!
! SIP UA configuration.
!
sip-ua
user-register
sip-username utda
sip-password utda
sip-server 192.168.0.3
register e164
!
!
! MGCP configuration.
!
mgcp
codec g711ulaw
vad
!
!
! Tones
voice class clear-down-tone 0 425 0 355 355
!
!
!
!
