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!
!
!
!
! VoIP configuration.
!
!
! Voice service voip configuration.
!
protocol sip
dtmf-relay out-of-band
voice service voip
fax protocol t38 redundancy 0
fax rate 9600
h323 call start fast
h323 call tunnel enable
no force-starth245
no force-h245address-at-setup
no call-barring unconfigured-ip-address
no voip-inbound-call-barring enable
!
!
! Voice port configuration.
!
! FXO
voice-port 0/0
connection plar 0004
caller-id enable
!
!
! FXO
voice-port 0/1
connection plar 0004
caller-id enable
shutdown
!
!
!
!
! Pots peer configuration.
!
dial-peer voice 100 pots
destination-pattern T
port 0/0
preference 9
!
dial-peer voice 101 pots
destination-pattern T
port 0/1
preference 9
!
!
!
! Voip peer configuration.
!
dial-peer voice 1000 voip
destination-pattern T
session target sip-server
session protocol sip
no vad
dtmf-relay dual-mode
!
dial-peer voice 1010 voip
destination-pattern T
session target ip ***.***.***.*** 5060
session protocol sip
no vad
dtmf-relay dual-mode
preference 1
!
!
!
!
dial-peer hunt 3
!
!
! Gateway configuration.
!
gateway
h323-id voip.***.***.***.***
signalling-port 1721
no ignore-msg-from-other-gk
!
!
! Codec classes configuration.
!
voice class codec 0
codec preference 1 g711ulaw
codec preference 2 g711alaw
codec preference 3 g729
codec preference 4 g7231r63
codec preference 5 g726r32
codec preference 6 g722r64
!
!
!
! SIP UA configuration.
!
sip-ua
signaling-port 5070
rport enable
no utf-encoded-format
register gateway
!
!
! MGCP configuration.
!
mgcp
codec g711ulaw
vad
!
!
! Tones
!
!
!
!
call-manager sip
ums-signaling-port 5062
reg-expire-value-fixed enable
!
!
call-manager sscp retry-counter 1
!
!
call-manager sscp store-event-time 3
call-manager sscp store-event-count 10
!
!
!
! CallManager service configuration.
!
!
!
!
!
!
!
call-manager cdr local on
!
!
ums mailbox qsize 300
!
!
ums smsreg chkduration 60
!
!
!
call-manager h323
signalling-port 1720
!
! Network Domain interface configuration.
!
network-domain interface ip FastEthernet0/0 domain public
network-domain interface ip FastEthernet0/1 domain private
!
rtp-proxy
idle-timeout 600
domain public
proxy address ip ***.***.***.***
proxy port-range ip 30000 50000
domain private
proxy address ip 192.168.3.254
proxy port-range ip 30000 50000
!
!
line console
!
line vty
!
character-set encoding usa ascii
! mount mem 1024 /tmp
mount mem 8192 /apcm
!
ldap
backup-dir /hd/ldap/backup
backup-timer 60
data-dir /hd/ldap
suffix "dc=addpac,dc=com"
rootdn "cn=Manager,dc=addpac,dc=com"
rootpw secret
include /hd/ldap/schema/core.schema
include /hd/ldap/schema/cosine.schema
include /hd/ldap/schema/inetorgperson.schema
include /hd/ldap/schema/addpac.schema
include /hd/ldap/schema/apcm.schema
include /hd/ldap/schema/apglobal.schema
include /hd/ldap/schema/apmessage.schema
include /hd/ldap/schema/apms.schema
include /hd/ldap/schema/apmd.schema
include /hd/ldap/schema/apums.schema
slapd 389
notification 5389
!
!
sqlitedbms
admin user root password ******
data-dir /hd/smartclient/database
dbms server
!
! presence server configuration.
!
presence
service enable
event-wait-time 10
sscp store-event-time 1
sscp store-event-count 1
!
!
! media server configuration.
!
media
rbt enable
!
!
! replication configuration.
!
rep
replication disable
!
!
ldapclient
host 127.0.0.1 389
ldap enable
!
!
!
sqlitedbms-client
dbserv 127.0.0.1 6543 ipnext.db
dbaccount user root password router
!
end

IP-PBX# [434874.905] VM(0/0/0) Rx FXO Ring Actv
[434874.905] VM(0/0/0) Tx RING_IND
6 <CEP 000000> : Call Received
[434876.655] VM(0/0/0) Rx FXO Ring Idle
[434876.655] VM(0/0/0) FXO input pass
[434876.655] VP(0/0/0) CallerId enable, std/gain 0/34
[434876.655] VP(0/0/0) open channel
[434876.655] VM(0/0/0) play mute
[434876.655] VP(0/0/0) Tx IBS signal 2/0
[434876.655] VP(0/0/0) Tx IBS dir 0
[434876.725] VP(0/0/0) GeneralEvent IBS gen end
[434879.655] VM(0/0/0) Ring Detect timeout
[434879.655] VM(0/0/0) FXO OnHook
[434879.655] VM(0/0/0) FXO input block
[434879.655] VM(0/0/0) vopp idle
[434879.655] VP(0/0/0) close channel
[434879.655] VM(0/0/0) Tx DISCONN_CNF
7 <CEP 000000> : Disconnected(16) at Busy
[434880.515] VM(0/0/0) Rx FXO Ring Actv
[434880.515] VM(0/0/0) Tx RING_IND
8 <CEP 000000> : Call Received
[434881.665] VM(0/0/0) Rx FXO Ring Idle
[434881.665] VM(0/0/0) FXO input pass
[434881.665] VP(0/0/0) CallerId enable, std/gain 0/34
[434881.665] VP(0/0/0) open channel
[434881.665] VM(0/0/0) play mute
[434881.665] VP(0/0/0) Tx IBS signal 2/0
[434881.665] VP(0/0/0) Tx IBS dir 0
[434881.730] VP(0/0/0) GeneralEvent IBS gen end
[434884.665] VM(0/0/0) Ring Detect timeout
[434884.665] VM(0/0/0) FXO OnHook
[434884.665] VM(0/0/0) FXO input block
[434884.665] VM(0/0/0) vopp idle
[434884.665] VP(0/0/0) close channel
[434884.665] VM(0/0/0) Tx DISCONN_CNF
9 <CEP 000000> : Disconnected(16) at Busy
[434885.525] VM(0/0/0) Rx FXO Ring Actv
[434885.525] VM(0/0/0) Tx RING_IND
10 <CEP 000000> : Call Received
[434886.675] VM(0/0/0) Rx FXO Ring Idle
[434886.675] VM(0/0/0) FXO input pass
[434886.675] VP(0/0/0) CallerId enable, std/gain 0/34
[434886.675] VP(0/0/0) open channel
[434886.675] VM(0/0/0) play mute
[434886.675] VP(0/0/0) Tx IBS signal 2/0
[434886.675] VP(0/0/0) Tx IBS dir 0
[434886.745] VP(0/0/0) GeneralEvent IBS gen end
[434889.675] VM(0/0/0) Ring Detect timeout
[434889.675] VM(0/0/0) FXO OnHook
[434889.675] VM(0/0/0) FXO input block
[434889.675] VM(0/0/0) vopp idle
[434889.675] VP(0/0/0) close channel
[434889.675] VM(0/0/0) Tx DISCONN_CNF
11 <CEP 000000> : Disconnected(16) at Busy
[434890.505] VM(0/0/0) Rx FXO Ring Actv
[434890.505] VM(0/0/0) Tx RING_IND
12 <CEP 000000> : Call Received
[434891.675] VM(0/0/0) Rx FXO Ring Idle
[434891.675] VM(0/0/0) FXO input pass
[434891.675] VP(0/0/0) CallerId enable, std/gain 0/34
[434891.675] VP(0/0/0) open channel
[434891.675] VM(0/0/0) play mute
[434891.675] VP(0/0/0) Tx IBS signal 2/0
[434891.675] VP(0/0/0) Tx IBS dir 0
[434891.745] VP(0/0/0) GeneralEvent IBS gen end

[435023.205] VM(0/0/0) Rx FXO Ring Actv
[435023.205] VM(0/0/0) Tx RING_IND
31 <CEP 000000> : Call Received
[435024.965] VM(0/0/0) Rx FXO Ring Idle
[435024.965] VM(0/0/0) FXO input pass
[435024.965] VP(0/0/0) CallerId enable, std/gain 0/34
[435024.965] VP(0/0/0) open channel
[435024.965] VM(0/0/0) play mute
[435024.965] VP(0/0/0) Tx IBS signal 2/0
[435024.965] VP(0/0/0) Tx IBS dir 0
[435025.030] VP(0/0/0) GeneralEvent IBS gen end
[435028.825] VM(0/0/0) Rx FXO Ring Actv
[435028.825] VM(0/0/0) vopp idle
[435028.825] VP(0/0/0) close channel
[435028.825] VM(0/0/0) FXO OnHook
[435028.825] VM(0/0/0) FXO input block
[435028.825] VM(0/0/0) FXO input block
[435028.825] VM(0/0/0) Tx OFFHOOK_IND
32 <CEP 000000> : Call Initiated : calledNumber() crv(0) total(0)
33 <Call 36> : ****** Call Created status(InitiatedByFXO) ver(8.50.013:Mar 14 2013) time(1374635045) ****
34 <CEP 000000> : Decode CID :
35 <CEP 000000> : Calling number()
36 <CEP 000000> : Call id(2544ef51-2db7-b373-815a-0002a4093728) callNum(36)
37 <Call 36> : MatchAllProcess After Sorted
<0> id(1000) dest(T) prefer(0) selected(20)
<1> id(1010) dest(T) prefer(1) selected(0)
<2> id(101) dest(T) prefer(9) selected(16)
<3> id(100) dest(T) prefer(9) selected(18)
38 <Call 36> : Initiate callee with dial-peer(T) status(CalleeDeterminedAll) id(2544ef51-2db7-b373-815a-0002a4093728)
39 <NetEP 36> : InitiateOutCall: calledNum(302) callingNum() target(sip-server)
40 <NetEP 36> : DoCall: calledAddr(sip:302@***.***.***.***:5060) callingAddr()
[435028.825] VM(0/0/0) set T38 enable by CCC
[435028.825] VM(0/0/0) set T38 mode STD
[435028.825] VM(0/0/0) Fax rate 9600
41 <SIP 36> : SetLocalAudioFormats : outbound(TRUE) hqaEnable(FALSE)
42 <SIP 36> : SetLocalAudioFormats : myVoipPeer is NULL, 999
43 <SIP 36> : SetLocalAudioFormats : outbound(TRUE) hqaEnable(FALSE)
44 <SIP 36> : SetLocalAudioFormats : myVoipPeer is NULL, 999
[435028.830] RTA(0/0/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0
G7222.2_AM 0
45 <SIP 0> : No authentication information available
46 <SIP 36> : Send INVITE Request

Sending SIP PDU to ( ***.***.***.***:5060 ) from 5070
INVITE sip:302@***.***.***.*** SIP/2.0
Via: SIP/2.0/UDP ***.***.***.***:5070;branch=z9hG4bK2551c95ba488
From: <sip:***.***.***.***:5070>;tag=2551c95ba4
To: <sip:302@***.***.***.***>
Call-ID: 2544ef51-76e8-c980-815b-0002a4093728@***.***.***.***
CSeq: 88 INVITE
Supported: replaces, timer, 100rel, early-session
Min-SE: 1800
Date: Wed, 24 Jul 2013 06:04:05 GMT
Session-Expires: 1800
User-Agent: AddPac SIP Gateway
Contact: <sip:***.***.***.***:5070>
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, PRACK, REFER, NOTIFY, INFO
Content-Type: application/sdp
Content-Length: 416
Max-Forwards: 70

v=0
o=- 1374635045 1374635045 IN IP4 ***.***.***.***
s=AddPac Gateway SDP
c=IN IP4 ***.***.***.***
t=1374635045 0
m=audio 23070 RTP/AVP 4 18 0 8 9 97 105 101
a=ptime:30
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:97 AMR-WB/8000
a=rtpmap:105 G7221/16000
a=fmtp:105 bitrate=32000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
[435028.850] RTA(0/0/0) Rx RS_LISTEN_REQ callId=36 ssId=1 G711U
peer=0.0.0.0 mp=23070/23071 hp=0/0

Received SIP PDU from ( ***.***.***.***:5060 )
SIP/2.0 100 Trying
Via: SIP/2.0/UDP ***.***.***.***:5070;branch=z9hG4bK2551c95ba488
From: <sip:***.***.***.***:5070>;tag=2551c95ba4
To: <sip:302@***.***.***.***>
Call-ID: 2544ef51-76e8-c980-815b-0002a4093728@***.***.***.***
CSeq: 88 INVITE
User-Agent: AddPac SIP Gateway
Content-Length: 0

47 <SIP 36> : Receive 100 Trying
48 <SIP 36> : Transaction (88 INVITE) proceeding

Received SIP PDU from ( ***.***.***.***:5060 )
SIP/2.0 200 OK
Via: SIP/2.0/UDP ***.***.***.***:5070;branch=z9hG4bK2551c95ba488
From: <sip:***.***.***.***:5070>;tag=2551c95ba4
To: <sip:302@***.***.***.***>;tag=2551ad5ca4
Call-ID: 2544ef51-76e8-c980-815b-0002a4093728@***.***.***.***
CSeq: 88 INVITE
Session-Expires: 1800;refresher=uas
User-Agent: AddPac SIP Gateway
Contact: <sip:302@***.***.***.***>
Content-Type: application/sdp
Content-Length: 227

v=0
o=addpac 1374635045 1374635045 IN IP4 ***.***.***.***
s=AddPac Gateway SDP
c=IN IP4 ***.***.***.***
t=1374635045 0
m=audio 26054 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

49 <SIP 36> : Receive 200 OK
50 <SIP 36> : Received INVITE OK response
51 <SIP 36> : Send ACK Request

Sending SIP PDU to ( ***.***.***.***:5060 ) from 5070
ACK sip:302@***.***.***.*** SIP/2.0
Via: SIP/2.0/UDP ***.***.***.***:5070;branch=z9hG4bK2551c95ba488
From: <sip:***.***.***.***:5070>;tag=2551c95ba4
To: <sip:302@***.***.***.***>;tag=2551ad5ca4
Call-ID: 2544ef51-76e8-c980-815b-0002a4093728@***.***.***.***
CSeq: 88 ACK
Content-Length: 0
Max-Forwards: 70

52 <SIP 36> : SetLocalAudioFormats : outbound(TRUE) hqaEnable(FALSE)
53 <SIP 36> : SetLocalAudioFormats : myVoipPeer is NULL, 999
54 <SIP 36> : Get SIP Audio MediaFormat : 0
[435028.920] RTA(0/0/0) Rx RS_OPEN_REQ callId=36 ssId=1 G711U
peer=***.***.***.*** mp=23070/23071 hp=26054/26055
55 <Call 36> : Connected from(fffffffe)
[435028.925] RTA(0/0/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0
VAD_CTRL 0
[435028.925] VM(0/0/0) VAD disable
[435028.925] VP(0/0/0) ignore notEnabledCh updating VAD 0
[435028.930] VM(0/0/0) SID enable by CCC
[435028.930] RTA(0/0/0) Rx CC_CONNECT_RSP peerId(0/0/0)
[435028.930] VM(0/0/0) FXO OffHook
[435028.930] VM(0/0/0) FXO input pass
[435028.930] VP(0/0/0) open channel
[435028.930] VP(0/0/0) attribute Fax enable, Modem disable
[435028.930] VP(0/0/0) update Fax enable, Modem disable
56 <SIP 283> : Receive ACK Request
57 <SIP 283> : Set Terminated Success for 88 INVITE
58 <NetEP 36> : Call with sip:302@***.***.***.*** established
[435028.935] RTA(0/0/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0
DTMF_CTRL 1
[435028.935] VM(0/0/0) DTMF_RTP_RFC2833 enable
[435028.935] RTA(0/0/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0
Rtp2833_DtmfPT TxPT=0x65 RxPT=0x65
[435028.935] VM(0/0/0) DTMF dual relay enable
59 <SIP 36> : Check Event Relation
60 <SIP 36> : Set Terminated Success for 88 INVITE
[435029.135] VM(0/0/0) Rx FXO Ring Idle
61 <SIP 283> : ReleaseWithBYE
62 <SIP 283> : Send BYE Request

Received SIP PDU from ( ***.***.***.***:5060 )
BYE sip:***.***.***.***:5070 SIP/2.0
Via: SIP/2.0/UDP ***.***.***.***:5060;branch=z9hG4bK2551ad5ca489
From: <sip:302@***.***.***.***>;tag=2551ad5ca4
To: <sip:***.***.***.***:5070>;tag=2551c95ba4
Call-ID: 2544ef51-76e8-c980-815b-0002a4093728@***.***.***.***
CSeq: 89 BYE
Date: Wed, 24 Jul 2013 06:04:09 GMT
User-Agent: AddPac SIP Gateway
Contact: <sip:302@***.***.***.***>
Content-Length: 0
Max-Forwards: 70

63 <SIP 36> : Receive BYE Request

Sending SIP PDU to ( ***.***.***.***:5060 ) from 5070
SIP/2.0 200 OK
Via: SIP/2.0/UDP ***.***.***.***:5060;branch=z9hG4bK2551ad5ca489
From: <sip:302@***.***.***.***>;tag=2551ad5ca4
To: <sip:***.***.***.***:5070>;tag=2551c95ba4
Call-ID: 2544ef51-76e8-c980-815b-0002a4093728@***.***.***.***
CSeq: 89 BYE
User-Agent: AddPac SIP Gateway
Content-Length: 0

64 <SIP 36> : ReleaseWithNothing
[435032.835] RTA(0/0/0) Rx RS_CLOSE_REQ callId=36 ssId=1 dir=all
[435032.835] RTA(0/0/0) close Media socket
[435032.835] RTA(0/0/0) close RTCP socket
65 <Call 36> : Terminated from(fffffffe) this(Remote:CallClear) before(<NULL>) forced(0) time(1374635049)
66 <CEP 000000> : DisconnectCall at Busy
67 <CEP 000000> : StopSignal
[435032.840] RTA(0/0/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0
DTMF_STOP
[435032.840] VM(0/0/0) play mute
[435032.840] VP(0/0/0) Tx IBS signal 2/0
[435032.840] VP(0/0/0) Tx IBS dir 0
68 <SIP 283> : Receive 200 OK
69 <SIP 283> : Transaction (89 BYE) completed
70 <CEP 000000> : Disconnect (0)
[435032.845] RTA(0/0/0) Rx CC_DISCONN_REQ CZ=0, peerId(0/0/0)
[435032.845] VM(0/0/0) vopp idle
[435032.845] VP(0/0/0) close channel
[435032.845] VM(0/0/0) FXO OnHook
[435032.845] VM(0/0/0) FXO input block
[435032.845] VM(0/0/0) Tx DISCONN_CNF
71 <NetEP 36> : Call TO <sip:302@***.***.***.***> terminated reason(Remote:CallClear)
72 <CEP 000000> : Disconnected(16) at Disconnecting

Автор:	JohnHaret [ 19 июл 2013, 10:44 ]
Заголовок сообщения:	IpNext50 FXO
Доброго времени суток, специалисты. Была приобретена IpNext50B с двумя FXO портами. Настройка шла по мануалам, благо их гора, но вот загвоздка одна осталась. Что имеем: Внутри телефоны - звонят. Звонок через Outgoing call rules работает. Звонок на сип транк и с сип транка (связь 2х АТС) работает. А вот внешний звонок на FXO порт дает отбой. в Incoming call rules одно правило: Trunks of Incoming Call -> Internal Trunk Gateway Single Extension Routing: Called Number Pattern T to extension 0004 partition internal Номер 0004 - это IVR. если из внутренней сети позвонить на него, слышно голос IVR сценария. вот вывод show run VOIP часть Код: ! ! ! ! ! VoIP configuration. ! ! ! Voice service voip configuration. ! protocol sip dtmf-relay out-of-band voice service voip fax protocol t38 redundancy 0 fax rate 9600 h323 call start fast h323 call tunnel enable no force-starth245 no force-h245address-at-setup no call-barring unconfigured-ip-address no voip-inbound-call-barring enable ! ! ! Voice port configuration. ! ! FXO voice-port 0/0 connection plar 0004 caller-id enable ! ! ! FXO voice-port 0/1 connection plar 0004 caller-id enable shutdown ! ! ! ! ! Pots peer configuration. ! dial-peer voice 100 pots destination-pattern T port 0/0 preference 9 ! dial-peer voice 101 pots destination-pattern T port 0/1 preference 9 ! ! ! ! Voip peer configuration. ! dial-peer voice 1000 voip destination-pattern T session target sip-server session protocol sip no vad dtmf-relay dual-mode ! dial-peer voice 1010 voip destination-pattern T session target ip *...* 5060 session protocol sip no vad dtmf-relay dual-mode preference 1 ! ! ! ! dial-peer hunt 3 ! ! ! Gateway configuration. ! gateway h323-id voip.*...* signalling-port 1721 no ignore-msg-from-other-gk ! ! ! Codec classes configuration. ! voice class codec 0 codec preference 1 g711ulaw codec preference 2 g711alaw codec preference 3 g729 codec preference 4 g7231r63 codec preference 5 g726r32 codec preference 6 g722r64 ! ! ! ! SIP UA configuration. ! sip-ua signaling-port 5070 rport enable no utf-encoded-format register gateway ! ! ! MGCP configuration. ! mgcp codec g711ulaw vad ! ! ! Tones ! ! ! ! call-manager sip ums-signaling-port 5062 reg-expire-value-fixed enable ! ! call-manager sscp retry-counter 1 ! ! call-manager sscp store-event-time 3 call-manager sscp store-event-count 10 ! ! ! ! CallManager service configuration. ! ! ! ! ! ! ! call-manager cdr local on ! ! ums mailbox qsize 300 ! ! ums smsreg chkduration 60 ! ! ! call-manager h323 signalling-port 1720 ! ! Network Domain interface configuration. ! network-domain interface ip FastEthernet0/0 domain public network-domain interface ip FastEthernet0/1 domain private ! rtp-proxy idle-timeout 600 domain public proxy address ip *...* proxy port-range ip 30000 50000 domain private proxy address ip 192.168.3.254 proxy port-range ip 30000 50000 ! ! line console ! line vty ! character-set encoding usa ascii ! mount mem 1024 /tmp mount mem 8192 /apcm ! ldap backup-dir /hd/ldap/backup backup-timer 60 data-dir /hd/ldap suffix "dc=addpac,dc=com" rootdn "cn=Manager,dc=addpac,dc=com" rootpw secret include /hd/ldap/schema/core.schema include /hd/ldap/schema/cosine.schema include /hd/ldap/schema/inetorgperson.schema include /hd/ldap/schema/addpac.schema include /hd/ldap/schema/apcm.schema include /hd/ldap/schema/apglobal.schema include /hd/ldap/schema/apmessage.schema include /hd/ldap/schema/apms.schema include /hd/ldap/schema/apmd.schema include /hd/ldap/schema/apums.schema slapd 389 notification 5389 ! ! sqlitedbms admin user root password ****** data-dir /hd/smartclient/database dbms server ! ! presence server configuration. ! presence service enable event-wait-time 10 sscp store-event-time 1 sscp store-event-count 1 ! ! ! media server configuration. ! media rbt enable ! ! ! replication configuration. ! rep replication disable ! ! ldapclient host 127.0.0.1 389 ldap enable ! ! ! sqlitedbms-client dbserv 127.0.0.1 6543 ipnext.db dbaccount user root password router ! end Заранее спасибо за помощь.

Автор:	JohnHaret [ 24 июл 2013, 03:05 ]
Заголовок сообщения:	Re: IpNext50 FXO
Так. Пока никто не отвечает, а лишь рассматривает мой пост. Что огорчает. Пока тестил, все ближе стал подбираться к корню проблемы. Если канал идет от AddPac 1100 FSX на IpNex50B FXO, то сигнал звонка на линию будет всегда занято. Так и не нашел сколько нужно выставить rind detect-timeout. Ставил 70, эфект не изменился. termonal monitor для rta ipc & voip call & voip sip молчит Взял линию, идущую от ГТС, и, о чудо, линия теперь не занята, идет стандартный ринг свободной, ожидание поднятия трубки. Но ничего не происходит. Вот дебаг Код: IP-PBX# [434874.905] VM(0/0/0) Rx FXO Ring Actv [434874.905] VM(0/0/0) Tx RING_IND 6 <CEP 000000> : Call Received [434876.655] VM(0/0/0) Rx FXO Ring Idle [434876.655] VM(0/0/0) FXO input pass [434876.655] VP(0/0/0) CallerId enable, std/gain 0/34 [434876.655] VP(0/0/0) open channel [434876.655] VM(0/0/0) play mute [434876.655] VP(0/0/0) Tx IBS signal 2/0 [434876.655] VP(0/0/0) Tx IBS dir 0 [434876.725] VP(0/0/0) GeneralEvent IBS gen end [434879.655] VM(0/0/0) Ring Detect timeout [434879.655] VM(0/0/0) FXO OnHook [434879.655] VM(0/0/0) FXO input block [434879.655] VM(0/0/0) vopp idle [434879.655] VP(0/0/0) close channel [434879.655] VM(0/0/0) Tx DISCONN_CNF 7 <CEP 000000> : Disconnected(16) at Busy [434880.515] VM(0/0/0) Rx FXO Ring Actv [434880.515] VM(0/0/0) Tx RING_IND 8 <CEP 000000> : Call Received [434881.665] VM(0/0/0) Rx FXO Ring Idle [434881.665] VM(0/0/0) FXO input pass [434881.665] VP(0/0/0) CallerId enable, std/gain 0/34 [434881.665] VP(0/0/0) open channel [434881.665] VM(0/0/0) play mute [434881.665] VP(0/0/0) Tx IBS signal 2/0 [434881.665] VP(0/0/0) Tx IBS dir 0 [434881.730] VP(0/0/0) GeneralEvent IBS gen end [434884.665] VM(0/0/0) Ring Detect timeout [434884.665] VM(0/0/0) FXO OnHook [434884.665] VM(0/0/0) FXO input block [434884.665] VM(0/0/0) vopp idle [434884.665] VP(0/0/0) close channel [434884.665] VM(0/0/0) Tx DISCONN_CNF 9 <CEP 000000> : Disconnected(16) at Busy [434885.525] VM(0/0/0) Rx FXO Ring Actv [434885.525] VM(0/0/0) Tx RING_IND 10 <CEP 000000> : Call Received [434886.675] VM(0/0/0) Rx FXO Ring Idle [434886.675] VM(0/0/0) FXO input pass [434886.675] VP(0/0/0) CallerId enable, std/gain 0/34 [434886.675] VP(0/0/0) open channel [434886.675] VM(0/0/0) play mute [434886.675] VP(0/0/0) Tx IBS signal 2/0 [434886.675] VP(0/0/0) Tx IBS dir 0 [434886.745] VP(0/0/0) GeneralEvent IBS gen end [434889.675] VM(0/0/0) Ring Detect timeout [434889.675] VM(0/0/0) FXO OnHook [434889.675] VM(0/0/0) FXO input block [434889.675] VM(0/0/0) vopp idle [434889.675] VP(0/0/0) close channel [434889.675] VM(0/0/0) Tx DISCONN_CNF 11 <CEP 000000> : Disconnected(16) at Busy [434890.505] VM(0/0/0) Rx FXO Ring Actv [434890.505] VM(0/0/0) Tx RING_IND 12 <CEP 000000> : Call Received [434891.675] VM(0/0/0) Rx FXO Ring Idle [434891.675] VM(0/0/0) FXO input pass [434891.675] VP(0/0/0) CallerId enable, std/gain 0/34 [434891.675] VP(0/0/0) open channel [434891.675] VM(0/0/0) play mute [434891.675] VP(0/0/0) Tx IBS signal 2/0 [434891.675] VP(0/0/0) Tx IBS dir 0 [434891.745] VP(0/0/0) GeneralEvent IBS gen end Выставил для FXO с ГТС rind detect-timeoute 70 и, опять чудо, логи: Код: [435023.205] VM(0/0/0) Rx FXO Ring Actv [435023.205] VM(0/0/0) Tx RING_IND 31 <CEP 000000> : Call Received [435024.965] VM(0/0/0) Rx FXO Ring Idle [435024.965] VM(0/0/0) FXO input pass [435024.965] VP(0/0/0) CallerId enable, std/gain 0/34 [435024.965] VP(0/0/0) open channel [435024.965] VM(0/0/0) play mute [435024.965] VP(0/0/0) Tx IBS signal 2/0 [435024.965] VP(0/0/0) Tx IBS dir 0 [435025.030] VP(0/0/0) GeneralEvent IBS gen end [435028.825] VM(0/0/0) Rx FXO Ring Actv [435028.825] VM(0/0/0) vopp idle [435028.825] VP(0/0/0) close channel [435028.825] VM(0/0/0) FXO OnHook [435028.825] VM(0/0/0) FXO input block [435028.825] VM(0/0/0) FXO input block [435028.825] VM(0/0/0) Tx OFFHOOK_IND 32 <CEP 000000> : Call Initiated : calledNumber() crv(0) total(0) 33 <Call 36> : **** Call Created status(InitiatedByFXO) ver(8.50.013:Mar 14 2013) time(1374635045) 34 <CEP 000000> : Decode CID : 35 <CEP 000000> : Calling number() 36 <CEP 000000> : Call id(2544ef51-2db7-b373-815a-0002a4093728) callNum(36) 37 <Call 36> : MatchAllProcess After Sorted <0> id(1000) dest(T) prefer(0) selected(20) <1> id(1010) dest(T) prefer(1) selected(0) <2> id(101) dest(T) prefer(9) selected(16) <3> id(100) dest(T) prefer(9) selected(18) 38 <Call 36> : Initiate callee with dial-peer(T) status(CalleeDeterminedAll) id(2544ef51-2db7-b373-815a-0002a4093728) 39 <NetEP 36> : InitiateOutCall: calledNum(302) callingNum() target(sip-server) 40 <NetEP 36> : DoCall: calledAddr(sip:302@...:5060) callingAddr() [435028.825] VM(0/0/0) set T38 enable by CCC [435028.825] VM(0/0/0) set T38 mode STD [435028.825] VM(0/0/0) Fax rate 9600 41 <SIP 36> : SetLocalAudioFormats : outbound(TRUE) hqaEnable(FALSE) 42 <SIP 36> : SetLocalAudioFormats : myVoipPeer is NULL, 999 43 <SIP 36> : SetLocalAudioFormats : outbound(TRUE) hqaEnable(FALSE) 44 <SIP 36> : SetLocalAudioFormats : myVoipPeer is NULL, 999 [435028.830] RTA(0/0/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0 G7222.2_AM 0 45 <SIP 0> : No authentication information available 46 <SIP 36> : Send INVITE Request Sending SIP PDU to ( ...:5060 ) from 5070 INVITE sip:302@... SIP/2.0 Via: SIP/2.0/UDP ...:5070;branch=z9hG4bK2551c95ba488 From: <sip:...:5070>;tag=2551c95ba4 To: <sip:302@...> Call-ID: 2544ef51-76e8-c980-815b-0002a4093728@... CSeq: 88 INVITE Supported: replaces, timer, 100rel, early-session Min-SE: 1800 Date: Wed, 24 Jul 2013 06:04:05 GMT Session-Expires: 1800 User-Agent: AddPac SIP Gateway Contact: <sip:...:5070> Accept: application/sdp Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, PRACK, REFER, NOTIFY, INFO Content-Type: application/sdp Content-Length: 416 Max-Forwards: 70 v=0 o=- 1374635045 1374635045 IN IP4 ... s=AddPac Gateway SDP c=IN IP4 ... t=1374635045 0 m=audio 23070 RTP/AVP 4 18 0 8 9 97 105 101 a=ptime:30 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:97 AMR-WB/8000 a=rtpmap:105 G7221/16000 a=fmtp:105 bitrate=32000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 [435028.850] RTA(0/0/0) Rx RS_LISTEN_REQ callId=36 ssId=1 G711U peer=0.0.0.0 mp=23070/23071 hp=0/0 Received SIP PDU from ( ...:5060 ) SIP/2.0 100 Trying Via: SIP/2.0/UDP ...:5070;branch=z9hG4bK2551c95ba488 From: <sip:...:5070>;tag=2551c95ba4 To: <sip:302@...> Call-ID: 2544ef51-76e8-c980-815b-0002a4093728@... CSeq: 88 INVITE User-Agent: AddPac SIP Gateway Content-Length: 0 47 <SIP 36> : Receive 100 Trying 48 <SIP 36> : Transaction (88 INVITE) proceeding Received SIP PDU from ( ...:5060 ) SIP/2.0 200 OK Via: SIP/2.0/UDP ...:5070;branch=z9hG4bK2551c95ba488 From: <sip:...:5070>;tag=2551c95ba4 To: <sip:302@...>;tag=2551ad5ca4 Call-ID: 2544ef51-76e8-c980-815b-0002a4093728@... CSeq: 88 INVITE Session-Expires: 1800;refresher=uas User-Agent: AddPac SIP Gateway Contact: <sip:302@...> Content-Type: application/sdp Content-Length: 227 v=0 o=addpac 1374635045 1374635045 IN IP4 ... s=AddPac Gateway SDP c=IN IP4 ... t=1374635045 0 m=audio 26054 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 49 <SIP 36> : Receive 200 OK 50 <SIP 36> : Received INVITE OK response 51 <SIP 36> : Send ACK Request Sending SIP PDU to ( ...:5060 ) from 5070 ACK sip:302@... SIP/2.0 Via: SIP/2.0/UDP ...:5070;branch=z9hG4bK2551c95ba488 From: <sip:...:5070>;tag=2551c95ba4 To: <sip:302@...>;tag=2551ad5ca4 Call-ID: 2544ef51-76e8-c980-815b-0002a4093728@... CSeq: 88 ACK Content-Length: 0 Max-Forwards: 70 52 <SIP 36> : SetLocalAudioFormats : outbound(TRUE) hqaEnable(FALSE) 53 <SIP 36> : SetLocalAudioFormats : myVoipPeer is NULL, 999 54 <SIP 36> : Get SIP Audio MediaFormat : 0 [435028.920] RTA(0/0/0) Rx RS_OPEN_REQ callId=36 ssId=1 G711U peer=... mp=23070/23071 hp=26054/26055 55 <Call 36> : Connected from(fffffffe) [435028.925] RTA(0/0/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0 VAD_CTRL 0 [435028.925] VM(0/0/0) VAD disable [435028.925] VP(0/0/0) ignore notEnabledCh updating VAD 0 [435028.930] VM(0/0/0) SID enable by CCC [435028.930] RTA(0/0/0) Rx CC_CONNECT_RSP peerId(0/0/0) [435028.930] VM(0/0/0) FXO OffHook [435028.930] VM(0/0/0) FXO input pass [435028.930] VP(0/0/0) open channel [435028.930] VP(0/0/0) attribute Fax enable, Modem disable [435028.930] VP(0/0/0) update Fax enable, Modem disable 56 <SIP 283> : Receive ACK Request 57 <SIP 283> : Set Terminated Success for 88 INVITE 58 <NetEP 36> : Call with sip:302@... established [435028.935] RTA(0/0/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0 DTMF_CTRL 1 [435028.935] VM(0/0/0) DTMF_RTP_RFC2833 enable [435028.935] RTA(0/0/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0 Rtp2833_DtmfPT TxPT=0x65 RxPT=0x65 [435028.935] VM(0/0/0) DTMF dual relay enable 59 <SIP 36> : Check Event Relation 60 <SIP 36> : Set Terminated Success for 88 INVITE [435029.135] VM(0/0/0) Rx FXO Ring Idle 61 <SIP 283> : ReleaseWithBYE 62 <SIP 283> : Send BYE Request Received SIP PDU from ( ...:5060 ) BYE sip:...:5070 SIP/2.0 Via: SIP/2.0/UDP ...:5060;branch=z9hG4bK2551ad5ca489 From: <sip:302@...>;tag=2551ad5ca4 To: <sip:...:5070>;tag=2551c95ba4 Call-ID: 2544ef51-76e8-c980-815b-0002a4093728@... CSeq: 89 BYE Date: Wed, 24 Jul 2013 06:04:09 GMT User-Agent: AddPac SIP Gateway Contact: <sip:302@...> Content-Length: 0 Max-Forwards: 70 63 <SIP 36> : Receive BYE Request Sending SIP PDU to ( ...:5060 ) from 5070 SIP/2.0 200 OK Via: SIP/2.0/UDP ...:5060;branch=z9hG4bK2551ad5ca489 From: <sip:302@...>;tag=2551ad5ca4 To: <sip:...:5070>;tag=2551c95ba4 Call-ID: 2544ef51-76e8-c980-815b-0002a4093728@... CSeq: 89 BYE User-Agent: AddPac SIP Gateway Content-Length: 0 64 <SIP 36> : ReleaseWithNothing [435032.835] RTA(0/0/0) Rx RS_CLOSE_REQ callId=36 ssId=1 dir=all [435032.835] RTA(0/0/0) close Media socket [435032.835] RTA(0/0/0) close RTCP socket 65 <Call 36> : Terminated from(fffffffe) this(Remote:CallClear) before(<NULL>) forced(0) time(1374635049) 66 <CEP 000000> : DisconnectCall at Busy 67 <CEP 000000> : StopSignal [435032.840] RTA(0/0/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0 DTMF_STOP [435032.840] VM(0/0/0) play mute [435032.840] VP(0/0/0) Tx IBS signal 2/0 [435032.840] VP(0/0/0) Tx IBS dir 0 68 <SIP 283> : Receive 200 OK 69 <SIP 283> : Transaction (89 BYE) completed 70 <CEP 000000> : Disconnect (0) [435032.845] RTA(0/0/0) Rx CC_DISCONN_REQ CZ=0, peerId(0/0/0) [435032.845] VM(0/0/0) vopp idle [435032.845] VP(0/0/0) close channel [435032.845] VM(0/0/0) FXO OnHook [435032.845] VM(0/0/0) FXO input block [435032.845] VM(0/0/0) Tx DISCONN_CNF 71 <NetEP 36> : Call TO <sip:302@...**> terminated reason(Remote:CallClear) 72 <CEP 000000> : Disconnected(16) at Disconnecting Но в отличии от остальных вариантов, АТС мне сообщает "Sory. You can't be connected right now." Что говорит о том, что вероятнее всего АТС не знает куда перевести звонок. Номер 302 присвоен SIP телефону, что стоит под рукой. Звонки с него и внутри сети на 302 работают.

Автор:	m.dark [ 30 июл 2013, 04:23 ]
Заголовок сообщения:	Re: IpNext50 FXO
Добавьте немного настроек в ваш конфиг: voice-port 0/0 connection plar 0004 ring detect-timeout 70 no comfort-noise fax-early-detect caller-id enable dial-peer voice 100 pots destination-pattern 9T port 0/0 preference 9 ! dial-peer voice 101 pots destination-pattern 9T port 0/1 preference 9 Дальше немного не понятно. Входящий вызов с FXO должен уходить на dial-peer voice 1000 voip destination-pattern T session target sip-server voice-class codec 0 session protocol sip no vad dtmf-relay dual-mode ! или на: dial-peer voice 1010 voip destination-pattern T session target ip *...* 5060 voice-class codec 0 session protocol sip no vad dtmf-relay dual-mode preference 1 Или тут так и должно быть? все разрулили preferenceами? Советую вам кодеки почистить, если вы знаете какие в вашей сети используются. voice class codec 0 codec preference 1 g729 codec preference 2 g711alaw codec preference 3 g711ulaw ваш вызов должен уходить на 0004 - это номер IVR, так ? Вызов на нее должен уходить через какой dial-peer voice XXXX voip? Вы пишите, что звоните с номера 302, только не пойму куда звоните? И каким образом должен по вашему пройти вызов? где зареган аппарат с номером 302? на этой же АТС и куда вы звоните? и что происходит?

Автор:	awsswa [ 31 июл 2013, 17:17 ]
Заголовок сообщения:	Re: IpNext50 FXO
что то у вас каша с настройками - вы по 323 что ли хотите общаться ?

Автор:	genal [ 01 авг 2013, 08:37 ]
Заголовок сообщения:	Re: IpNext50 FXO
Это IP АТС а не шлюз. пирами тут все не разрулить. Никакой каши нет.

СвязьПроект http://old.xdsl.ru/svpro/

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