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Asterisk и Addpac AP1000 не проходят входящие звонки
http://old.xdsl.ru/svpro/viewtopic.php?f=4&t=5588		 Страница 1 из 1
Автор:  Taipan.GAV [ 04 сен 2017, 09:29 ]
Заголовок сообщения:  Asterisk и Addpac AP1000 не проходят входящие звонки
Доброго времени суток, уважаемые!
Подскажите что я делаю не так?
Есть связка asterisk + addpack ap1000. Исходящие звонки проходят без проблем.
Входящие на софт-фоны проходят, а на Addpac не идут.
В show sip пишет, что нет регистрации:
Код:
AP1000# show sip

Proxyserver Registration Information
   proxyserver registration option = e164
   Proxyserver list :
      ---------------------------------------------------------------
         Server address           Port    Priority         Status
      ---------------------------------------------------------------
        192.168.0.248             5060      128       Not Registered

   Proxyserver registration status :
      -----------------------------------------------------------------------------------
         UserName          UserID            Password            Port           Status
      -----------------------------------------------------------------------------------
         200               200               f9002b190a745e37e902072a829ecbea     0/ 0      Not Registered
         201               201               664447627b85863f5e46ad7136ed0ad7     0/ 1      Not Registered
         202               202               7732353c84f6e65eee3abe4b1047c621     0/ 2      Not Registered
         203               203               d080980eff581245cf17dcebc7450742     0/ 3      Not Registered

SIP UA Timer counters
   retry counter = 10
SIP UA Timer values
   tretry (sip retry timer) = 500 msec.
   tinterval (sip retry max interval timer) = 4 sec.
   treg (sip register timer) = 3600 sec.
   tregtry (sip register retry timer) = 20 sec.
   texpires (sip invite expire timer) = 180 sec.
   tsipping (sip ping timer) = 45 sec.
SIP UA MIN-SE value
   Min-SE = 1800 sec.
SIP DNS SRV Query : Disable
SIP Call Transfer Mode : Basic
SIP Media Channel Start Mode : Default
SIP Reliable Provisional Response Option : Disabled
SIP Response Option : default
SIP Local Domain : NULL
Special Char : NULL
SIP Routing Method of Incoming Call : Default
SIP Remote-Party-ID : Disabled
SIP Local Host Name : No
SIP Conference Server Info
   Name (ID) = NULL
   Related Voip Tag = -1
SIP NAT Info
   PING = Disabled
   Required = NULL
AP1000#


Конфиг Addpac`а на данный момент такой:
Код:
!
version 8.30K
!
hostname AP1000
!
!
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
!
!
ip-share enable
ip-share interface net-side ether0.0
ip-share interface local-side ether1.0
!
interface ether0.0
 ip address 192.168.0.247 255.255.255.0
!
interface ether1.0
 no ip address
!
snmp name AP1000
!
no arp reset
!
route 0.0.0.0 0.0.0.0 192.168.0.254
!
!
!
!
!
! 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
 busyout monitor gatekeeper
 busyout monitor sip-server
-- more --
 no busyout monitor callagent
 busyout monitor voip-interface
!
!
! Voice port configuration.
!
! FXS
voice-port 0/0
 input gain 1
 output gain 1
 no comfort-noise
 fax-early-detect
 no announcement
 caller-id enable
!
!
! FXS
voice-port 0/1
 input gain 1
 output gain 1
 no comfort-noise
 fax-early-detect
 no announcement
 caller-id enable
!
!
! FXS
voice-port 0/2
 input gain 1
 output gain 1
 no comfort-noise
 fax-early-detect
 no announcement
 caller-id enable
!
!
! FXS
voice-port 0/3
 input gain 1
 output gain 1
 no comfort-noise
 fax-early-detect
 no announcement
 caller-id enable
!
!
!
!
! Pots peer configuration.
!
dial-peer voice 0 pots
 destination-pattern 200
 port 0/0
 user-password f9002b190a745e37e902072a829ecbea
!
dial-peer voice 1 pots
 destination-pattern 201
 port 0/1
 user-password 664447627b85863f5e46ad7136ed0ad7
!
dial-peer voice 2 pots
 destination-pattern 202
 port 0/2
 user-password 7732353c84f6e65eee3abe4b1047c621
!
dial-peer voice 3 pots
 destination-pattern 203
 port 0/3
 user-password d080980eff581245cf17dcebc7450742
!
!
!
! Voip peer configuration.
!
dial-peer voice 1001 voip
 destination-pattern .T
 session target 192.168.0.248
 session protocol sip
 voice-class codec 0
 dtmf-relay rtp-2833
 no vad
!
!
!
!
!
!
! Gateway configuration.
!
gateway
 h323-id voip.192.168.0.247
 no ignore-msg-from-other-gk
!
!
! Codec classes configuration.
!
voice class codec 0
 codec preference 1 g729
 codec preference 2 g711alaw
 codec preference 3 g711ulaw
!
!
!
! SIP UA configuration.
!
sip-ua
 user-register
 sip-server 192.168.0.248
 timeout treg 3600
 register e164
!
!
! MGCP configuration.
!
mgcp
 codec g711ulaw
 vad
!
!
! Tones
!
!
!
!

Дело в том, что я впервые вижу данную железяку, до сего момента с Addpac`ами дел не имел.
Автор:  Taipan.GAV [ 04 сен 2017, 15:08 ]
Заголовок сообщения:  Re: Asterisk и Addpac AP1000 не проходят входящие звонки
Вот так выглядит debug voip call при попытке позвонить с софт-фона на addpac
Код:
AP1000# 1       <Call   63>     : ******************  Call Created status(InitiatedByNet)  *******************
2       <SIP    63>     : Receive INVITE Request
3       <NetCon 63>     : Found inbound voip peer by dest-pattern id(1001)
4       <Call   63>     : From Net - calledParty() callingParty(102)
5       <Call   63>     : Terminated  from(fffffff7) this(Local:InvalidNumber) before(NULL) forced(0)
6       <NetEP  63>     : Call TO <102> terminated reason(Local:InvalidNumber)
7       <SIP    63>     : Receive ACK Request
8       <SIP    63>     : Set Terminated Success for 102 INVITE


А так выглядит debug voip sip при тех же условиях:
Код:

        Received SIP PDU from ( 192.168.0.248:5060 )
INVITE sip:192.168.0.247:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.248:5060;branch=z9hG4bK764ed2b4;rport
Max-Forwards: 70
From: "102" <sip:102@192.168.0.248>;tag=as6edffcff
To: <sip:192.168.0.247:5060>
Contact: <sip:102@192.168.0.248:5060>
Call-ID: 3d6c178810453c2c0a1a893463ff4254@192.168.0.248:5060
CSeq: 102 INVITE
User-Agent: FPBX-13.0.192.16(13.15.0)
Date: Mon, 04 Sep 2017 12:59:34 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
P-Asserted-Identity: "102" <sip:102@192.168.0.248>
Content-Type: application/sdp
Content-Length: 332

v=0
o=root 1959644933 1959644933 IN IP4 192.168.0.248
s=Asterisk PBX 13.15.0
c=IN IP4 192.168.0.248
t=0 0
m=audio 27424 RTP/AVP 0 8 3 111 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

        Sending SIP PDU to ( 192.168.0.248:5060 ) from 5060
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.248:5060;branch=z9hG4bK764ed2b4;rport
From: "102" <sip:102@192.168.0.248>;tag=as6edffcff
To: <sip:192.168.0.247:5060>
Call-ID: 3d6c178810453c2c0a1a893463ff4254@192.168.0.248:5060
CSeq: 102 INVITE
User-Agent: AddPac SIP Gateway
Content-Length: 0



        Sending SIP PDU to ( 192.168.0.248:5060 ) from 5060
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.0.248:5060;branch=z9hG4bK764ed2b4;rport
From: "102" <sip:102@192.168.0.248>;tag=as6edffcff
To: <sip:192.168.0.247:5060>;tag=8559e46ea4
Call-ID: 3d6c178810453c2c0a1a893463ff4254@192.168.0.248:5060
CSeq: 102 INVITE
User-Agent: AddPac SIP Gateway
Content-Length: 0



        Received SIP PDU from ( 192.168.0.248:5060 )
ACK sip:192.168.0.247:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.248:5060;branch=z9hG4bK764ed2b4;rport
Max-Forwards: 70
From: "102" <sip:102@192.168.0.248>;tag=as6edffcff
To: <sip:192.168.0.247:5060>;tag=8559e46ea4
Contact: <sip:102@192.168.0.248:5060>
Call-ID: 3d6c178810453c2c0a1a893463ff4254@192.168.0.248:5060
CSeq: 102 ACK
User-Agent: FPBX-13.0.192.16(13.15.0)
Content-Length: 0
Автор:  Taipan.GAV [ 05 сен 2017, 13:29 ]
Заголовок сообщения:  Re: Asterisk и Addpac AP1000 не проходят входящие звонки
Debug voip sip при звонке с Addpac`а на софт-фон
Код:
        Sending SIP PDU to ( 192.168.0.248:5060 ) from 5060
INVITE sip:102@192.168.0.248 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.247:5060;branch=z9hG4bKe9593b28a470
From: <sip:202@192.168.0.247>;tag=e9593b28a4
To: <sip:102@192.168.0.248>
Call-ID: e9d1ae59-9dc1-3ba5-8128-0002a4034602@192.168.0.247
CSeq: 70 INVITE
Supported: timer, replaces
Min-SE: 1800
Date: Tue, 05 Sep 2017 16:33:45 GMT
User-Agent: AddPac SIP Gateway
Contact: <sip:202@192.168.0.247>
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, REFER, NOTIFY, INFO
Content-Type: application/sdp
Content-Length: 282
Max-Forwards: 70

v=0
o=202 1504629225 1504629225 IN IP4 192.168.0.247
s=AddPac Gateway SDP
c=IN IP4 192.168.0.247
t=1504629225 0
m=audio 23402 RTP/AVP 18 8 0 101
a=rtpmap:18 G729/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20


        Received SIP PDU from ( 192.168.0.248:5060 )
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.247:5060;branch=z9hG4bKe9593b28a470;received=192.168.0.247;rport=5060
From: <sip:202@192.168.0.247>;tag=e9593b28a4
To: <sip:102@192.168.0.248>;tag=as753923f8
Call-ID: e9d1ae59-9dc1-3ba5-8128-0002a4034602@192.168.0.247
CSeq: 70 INVITE
Server: FPBX-13.0.192.16(13.15.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3e757d49"
Content-Length: 0


        Sending SIP PDU to ( 192.168.0.248:5060 ) from 5060
ACK sip:102@192.168.0.248 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.247:5060;branch=z9hG4bKe9593b28a470
From: <sip:202@192.168.0.247>;tag=e9593b28a4
To: <sip:102@192.168.0.248>;tag=as753923f8
Call-ID: e9d1ae59-9dc1-3ba5-8128-0002a4034602@192.168.0.247
CSeq: 70 ACK
Content-Length: 0
Max-Forwards: 70



        Sending SIP PDU to ( 192.168.0.248:5060 ) from 5060
INVITE sip:102@192.168.0.248 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.247:5060;branch=z9hG4bKe9593b28a471
From: <sip:202@192.168.0.247>;tag=e9593b28a4
To: <sip:102@192.168.0.248>
Call-ID: e9d1ae59-9dc1-3ba5-8128-0002a4034602@192.168.0.247
CSeq: 71 INVITE
Supported: timer, replaces
Min-SE: 1800
Date: Tue, 05 Sep 2017 16:33:45 GMT
User-Agent: AddPac SIP Gateway
Authorization: Digest username="202", realm="asterisk", nonce="3e757d49", uri="sip:102@192.168.0.248", response="c63be5ccf2e89378fabb6e9c83c68cf2", algorithm=MD5
Contact: <sip:202@192.168.0.247>
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, REFER, NOTIFY, INFO
Content-Type: application/sdp
Content-Length: 290
Max-Forwards: 70

v=0
o=202 1504629225 1504629225 IN IP4 192.168.0.247
s=AddPac Gateway SDP
c=IN IP4 192.168.0.247
t=1504629225 0
m=audio 23402 RTP/AVP 18 8 0 101
a=rtpmap:18 G729/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-15
a=ptime:20


        Received SIP PDU from ( 192.168.0.248:5060 )
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.247:5060;branch=z9hG4bKe9593b28a471;received=192.168.0.247;rport=5060
From: <sip:202@192.168.0.247>;tag=e9593b28a4
To: <sip:102@192.168.0.248>
Call-ID: e9d1ae59-9dc1-3ba5-8128-0002a4034602@192.168.0.247
CSeq: 71 INVITE
Server: FPBX-13.0.192.16(13.15.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:102@192.168.0.248:5060>
Content-Length: 0


        Received SIP PDU from ( 192.168.0.248:5060 )
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.247:5060;branch=z9hG4bKe9593b28a471;received=192.168.0.247;rport=5060
From: <sip:202@192.168.0.247>;tag=e9593b28a4
To: <sip:102@192.168.0.248>;tag=as760bb0c5
Call-ID: e9d1ae59-9dc1-3ba5-8128-0002a4034602@192.168.0.247
CSeq: 71 INVITE
Server: FPBX-13.0.192.16(13.15.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:102@192.168.0.248:5060>
P-Asserted-Identity: "102" <sip:102@192.168.0.247>
Content-Length: 0


        Received SIP PDU from ( 192.168.0.248:5060 )
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.247:5060;branch=z9hG4bKe9593b28a471;received=192.168.0.247;rport=5060
From: <sip:202@192.168.0.247>;tag=e9593b28a4
To: <sip:102@192.168.0.248>;tag=as760bb0c5
Call-ID: e9d1ae59-9dc1-3ba5-8128-0002a4034602@192.168.0.247
CSeq: 71 INVITE
Server: FPBX-13.0.192.16(13.15.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:102@192.168.0.248:5060>
Content-Length: 0


        Sending SIP PDU to ( 192.168.0.248:5060 ) from 5060
CANCEL sip:102@192.168.0.248 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.247:5060;branch=z9hG4bKe9593b28a471
From: <sip:202@192.168.0.247>;tag=e9593b28a4
To: <sip:102@192.168.0.248>
Call-ID: e9d1ae59-9dc1-3ba5-8128-0002a4034602@192.168.0.247
CSeq: 71 CANCEL
Date: Tue, 05 Sep 2017 16:33:53 GMT
User-Agent: AddPac SIP Gateway
Content-Length: 0
Max-Forwards: 70



        Received SIP PDU from ( 192.168.0.248:5060 )
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.0.247:5060;branch=z9hG4bKe9593b28a471;received=192.168.0.247;rport=5060
From: <sip:202@192.168.0.247>;tag=e9593b28a4
To: <sip:102@192.168.0.248>;tag=as760bb0c5
Call-ID: e9d1ae59-9dc1-3ba5-8128-0002a4034602@192.168.0.247
CSeq: 71 INVITE
Server: FPBX-13.0.192.16(13.15.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


        Sending SIP PDU to ( 192.168.0.248:5060 ) from 5060
ACK sip:102@192.168.0.248 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.247:5060;branch=z9hG4bKe9593b28a471
From: <sip:202@192.168.0.247>;tag=e9593b28a4
To: <sip:102@192.168.0.248>;tag=as760bb0c5
Call-ID: e9d1ae59-9dc1-3ba5-8128-0002a4034602@192.168.0.247
CSeq: 71 ACK
Content-Length: 0
Max-Forwards: 70



        Received SIP PDU from ( 192.168.0.248:5060 )
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.247:5060;branch=z9hG4bKe9593b28a471;received=192.168.0.247;rport=5060
From: <sip:202@192.168.0.247>;tag=e9593b28a4
To: <sip:102@192.168.0.248>;tag=as760bb0c5
Call-ID: e9d1ae59-9dc1-3ba5-8128-0002a4034602@192.168.0.247
CSeq: 71 CANCEL
Server: FPBX-13.0.192.16(13.15.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0



А это Debug voip call при звонке внутри Addpac`а с одного порта на другой
Код:
1       <CEP    000200> : Call Received
2       <CEP    000200> : Call Initiated : calledNumber() crv(0) total(0)
3       <Call   220>    : ******************  Call Created status(InitiatedByFXS)  *******************
4       <CEP    000200> : Calling number(202)
5       <CEP    000200> : Call id(95d2ae59-346d-25ff-8130-0002a4034602) callNum(220)
6       <Call   220>    : Digit(2) at InitiatedByFXS
7       <Call   220>    : MatchedAll
8       <Call   220>    : Digit(0) at CalleeDeterminedWaitDigit
9       <Call   220>    : MatchedAll
10      <Call   220>    : Digit(3) at CalleeDeterminedWaitDigit
11      <Call   220>    : MatchedPerfect
12      <Call   220>    : MatchAllProcess After Sorted
                          <0>  id(3) dest(203) prefer(0) selected(2)
                          <1>  id(1001) dest(.T) prefer(0) selected(29)
13      <Call   220>    : Initiate callee with dial-peer(203) status(CalleeDeterminedAll) id(95d2ae59-346d-25ff-8130-0002a4034602)
14      <CEP    000300> : InitiateOutCall :  calledNum(), callingNum(202), callerPort(200) type(FXS)
15      <CEP    000300> : Outbound call to CEP callId(95d2ae59-346d-25ff-8130-0002a4034602) callNum(220)
16      <Call   220>    : Connected from(300)
17      <Call   220>    : Connected from(200)
18      <CEP    000200> : Disconnected(16) at Busy
19      <Call   220>    : Terminated  from(200) this(Local:CallClear) before(NULL) forced(0)
20      <CEP    000200> : DisconnectCall at Idle
21      <CEP    000300> : DisconnectCall at Busy
22      <CEP    000300> : StopSignal
23      <CEP    000300> : Disconnect (0)
24      <CEP    000300> : Disconnected(16) at Disconnecting
Автор:  Taipan.GAV [ 05 сен 2017, 13:41 ]
Заголовок сообщения:  Re: Asterisk и Addpac AP1000 не проходят входящие звонки
Прописал на Asterisk`е каждому порту Addpac`а (они выступают в качестве extentions) IP-адреса. После чего они стали регистрироваться на астериске, но аддпак по прежнему пишет, что нет регистрации.

Код:
Name/username             Host                                    Dyn Forcerport Comedia    ACL Port     Status      Description
101                       (Unspecified)                            D  Yes        Yes         A  0        UNKNOWN
102/102                   192.168.0.101                            D  Yes        Yes         A  63546    OK (111 ms)
123/123                   188.18.112.64                            D  Yes        Yes         A  54066    OK (64 ms)
200                       192.168.0.247                               Yes        Yes         A  5060     OK (37 ms)
201                       192.168.0.247                               Yes        Yes         A  5060     OK (37 ms)
202                       192.168.0.247                               Yes        Yes         A  5060     OK (37 ms)
203                       192.168.0.247                               Yes        Yes         A  5060     OK (37 ms)
7XXXXXXXXXX/7XXXXXXXXXXID IP addres                                   Yes        Yes            5060     OK (5 ms)
7XXXXXXXXXX/7XXXXXXXXXXID IP addres                                   Yes        Yes            5060     OK (6 ms)
999                       (Unspecified)                            D  Yes        Yes         A  0        UNKNOWN
10 sip peers [Monitored: 8 online, 2 offline Unmonitored: 0 online, 0 offline]
Автор:  genal [ 28 сен 2017, 10:45 ]
Заголовок сообщения:  Re: Asterisk и Addpac AP1000 не проходят входящие звонки
У Вас почему то в логах не видно запроса регистрации на адпаке. Посмотрите, вообще шлет ли он регистрацию. Возможно нужно переписать конфиг заново, или поменять прошивку (можно попробовать даже на ту же самую). В sip-ua попробуйте добваить remote-party-id.
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