Alcatel <-> Avaya <sip> FPBX-2.11 проблема с сотовыми телефонами

Приветствую!

Есть проблемка.

Если позвонить с городского телефона или внутреннего подключенного к avaya или Asterisk то проблем нет.
Если позвонить с сотового, то при поднятии трубки вызов скидывается.

ASTERISK


sip set debug peer 4970
SIP Debugging Enabled for IP: 10.214.49.183
Really destroying SIP dialog '67a04499-1735f94c@10.214.49.183' Method: SUBSCRIBE
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
Audio is at 19410
Adding codec 100004 (alaw) to SDP
Adding codec 100003 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.214.49.183:5062:
INVITE sip:4970@10.214.49.183:5062;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.214.128.218:5060;branch=z9hG4bK7ec612d1
Max-Forwards: 70
From: <sip:89144149111@10.214.128.218>;tag=as62cf889b
To: <sip:4970@10.214.49.183:5062;transport=tcp>
Contact: <sip:89144149111@10.214.128.218:5060;transport=tcp>
Call-ID: 4cbb6a88678efe261a047f6e235a2085@10.214.128.218:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.11.0beta2(11.2.1)
Date: Wed, 26 Apr 2017 02:04:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 263
 v=0
o=root 1377381636 1377381636 IN IP4 10.214.128.218
s=Asterisk PBX 11.2.1
c=IN IP4 10.214.128.218
t=0 0
m=audio 19410 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
 
 <--- SIP read from TCP:10.214.49.183:5062 --->
SIP/2.0 401 Unauthorized
To: <sip:4970@10.214.49.183:5062>;tag=e4660d33fc3ddea4i0
From: <sip:89144149111@10.214.128.218>;tag=as62cf889b
Call-ID: 4cbb6a88678efe261a047f6e235a2085@10.214.128.218:5060
CSeq: 102 INVITE
Via: SIP/2.0/TCP 10.214.128.218:5060;branch=z9hG4bK7ec612d1
Server: Cisco/SPA504G-7.5.6(XU)
WWW-Authenticate: Digest realm="10.214.128.218", nonce="f67afa12", qop="auth", algorithm=md5
Content-Length: 0
 <------------->
--- (9 headers 0 lines) ---
Transmitting (no NAT) to 10.214.49.183:5062:
ACK sip:4970@10.214.49.183:5062;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.214.128.218:5060;branch=z9hG4bK7ec612d1
Max-Forwards: 70
From: <sip:89144149111@10.214.128.218>;tag=as62cf889b
To: <sip:4970@10.214.49.183:5062;transport=tcp>;tag=e4660d33fc3ddea4i0
Contact: <sip:89144149111@10.214.128.218:5060;transport=tcp>
Call-ID: 4cbb6a88678efe261a047f6e235a2085@10.214.128.218:5060
CSeq: 102 ACK
User-Agent: FPBX-2.11.0beta2(11.2.1)
Content-Length: 0
 
 Audio is at 19410
Adding codec 100004 (alaw) to SDP
Adding codec 100003 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.214.49.183:5062:
INVITE sip:4970@10.214.49.183:5062;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.214.128.218:5060;branch=z9hG4bK33e71a6d
Max-Forwards: 70
From: <sip:89144149111@10.214.128.218>;tag=as62cf889b
To: <sip:4970@10.214.49.183:5062;transport=tcp>
Contact: <sip:89144149111@10.214.128.218:5060;transport=tcp>
Call-ID: 4cbb6a88678efe261a047f6e235a2085@10.214.128.218:5060
CSeq: 103 INVITE
User-Agent: FPBX-2.11.0beta2(11.2.1)
Authorization: Digest username="4970", realm="10.214.128.218", algorithm=MD5, uri="sip:4970@10.214.49.183:5062;transport=tcp", nonce="f67afa12", response="79984e0d33a5ac58ed3702b3f6e5fe3e", qop=auth, cnonce="5c08bb00", nc=00000001
Date: Wed, 26 Apr 2017 02:04:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 263
 v=0
o=root 1377381636 1377381637 IN IP4 10.214.128.218
s=Asterisk PBX 11.2.1
c=IN IP4 10.214.128.218
t=0 0
m=audio 19410 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
 
 <--- SIP read from TCP:10.214.49.183:5062 --->
SIP/2.0 100 Trying
To: <sip:4970@10.214.49.183:5062>
From: <sip:89144149111@10.214.128.218>;tag=as62cf889b
Call-ID: 4cbb6a88678efe261a047f6e235a2085@10.214.128.218:5060
CSeq: 103 INVITE
Via: SIP/2.0/TCP 10.214.128.218:5060;branch=z9hG4bK33e71a6d
Server: Cisco/SPA504G-7.5.6(XU)
Content-Length: 0
 <------------->
--- (8 headers 0 lines) ---
 <--- SIP read from TCP:10.214.49.183:5062 --->
SIP/2.0 180 Ringing
To: <sip:4970@10.214.49.183:5062>;tag=d4ad9f77c7c63156i0
From: <sip:89144149111@10.214.128.218>;tag=as62cf889b
Call-ID: 4cbb6a88678efe261a047f6e235a2085@10.214.128.218:5060
CSeq: 103 INVITE
Via: SIP/2.0/TCP 10.214.128.218:5060;branch=z9hG4bK33e71a6d
Contact: "4970" <sip:4970@10.214.49.183:5062;transport=tcp>
Server: Cisco/SPA504G-7.5.6(XU)
Content-Length: 0
 <------------->
--- (9 headers 0 lines) ---
list_route: hop: <sip:4970@10.214.49.183:5062;transport=tcp>
 <--- SIP read from TCP:10.214.49.183:5062 --->
SIP/2.0 200 OK
To: <sip:4970@10.214.49.183:5062>;tag=d4ad9f77c7c63156i0
From: <sip:89144149111@10.214.128.218>;tag=as62cf889b
Call-ID: 4cbb6a88678efe261a047f6e235a2085@10.214.128.218:5060
CSeq: 103 INVITE
Via: SIP/2.0/TCP 10.214.128.218:5060;branch=z9hG4bK33e71a6d
Contact: "4970" <sip:4970@10.214.49.183:5062;transport=tcp>
Server: Cisco/SPA504G-7.5.6(XU)
Content-Length: 212
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE
Supported: replaces
Content-Type: application/sdp
 v=0
o=- 15101922 15101922 IN IP4 10.214.49.183
s=-
c=IN IP4 10.214.49.183
t=0 0
m=audio 16464 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
<------------->
--- (12 headers 11 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 10.214.49.183:16464
listroute: hop: <sip:4970@10.214.49.183:5062;transport=tcp>
setdestination: Parsing <sip:4970@10.214.49.183:5062;transport=tcp> for address/port to send to
set_destination: set destination to 10.214.49.183:5062
Transmitting (no NAT) to 10.214.49.183:5062:
ACK sip:4970@10.214.49.183:5062;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.214.128.218:5060;branch=z9hG4bK3804194c
Max-Forwards: 70
From: <sip:89144149111@10.214.128.218>;tag=as62cf889b
To: <sip:4970@10.214.49.183:5062;transport=tcp>;tag=d4ad9f77c7c63156i0
Contact: <sip:89144149111@10.214.128.218:5060;transport=tcp>
Call-ID: 4cbb6a88678efe261a047f6e235a2085@10.214.128.218:5060
CSeq: 103 ACK
User-Agent: FPBX-2.11.0beta2(11.2.1)
Content-Length: 0
 
 == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'SIP/avayaoutgoing-000000a9' in macro 'hangupcall'
  == Spawn extension (macro-dial-one, h, 1) exited non-zero on 'SIP/avayaoutgoing-000000a9'
Scheduling destruction of SIP dialog '4cbb6a88678efe261a047f6e235a2085@10.214.128.218:5060' in 6400 ms (Method: INVITE)
setdestination: Parsing <sip:4970@10.214.49.183:5062;transport=tcp> for address/port to send to
setdestination: set destination to 10.214.49.183:5062
Reliably Transmitting (no NAT) to 10.214.49.183:5062:
BYE sip:4970@10.214.49.183:5062;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.214.128.218:5060;branch=z9hG4bK5f5e1d29
Max-Forwards: 70
From: <sip:89144149111@10.214.128.218>;tag=as62cf889b
To: <sip:4970@10.214.49.183:5062;transport=tcp>;tag=d4ad9f77c7c63156i0
Call-ID: 4cbb6a88678efe261a047f6e235a2085@10.214.128.218:5060
CSeq: 104 BYE
User-Agent: FPBX-2.11.0beta2(11.2.1)
Authorization: Digest username="4970", realm="10.214.128.218", algorithm=MD5, uri="sip:4970@10.214.49.183:5062", nonce="f67afa12", response="5d1ac76774f669765e745c9807bd3987", qop=auth, cnonce="63751bfa", nc=00000002
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
 
 == Spawn extension (macro-dial-one, s, 42) exited non-zero on 'SIP/avayaoutgoing-000000a9' in macro 'dial-one'
  == Spawn extension (macro-exten-vm, s, 9) exited non-zero on 'SIP/avayaoutgoing-000000a9' in macro 'exten-vm'
  == Spawn extension (from-trunk, 4970, 2) exited non-zero on 'SIP/avaya_outgoing-000000a9'
 <--- SIP read from TCP:10.214.49.183:5062 --->
SIP/2.0 200 OK
To: <sip:4970@10.214.49.183:5062>;tag=d4ad9f77c7c63156i0
From: <sip:89144149111@10.214.128.218>;tag=as62cf889b
Call-ID: 4cbb6a88678efe261a047f6e235a2085@10.214.128.218:5060
CSeq: 104 BYE
Via: SIP/2.0/TCP 10.214.128.218:5060;branch=z9hG4bK5f5e1d29
Server: Cisco/SPA504G-7.5.6(XU)
Content-Length: 0
 <------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '4cbb6a88678efe261a047f6e235a2085@10.214.128.218:5060' Method: INVITE
localhost*CLI> sip set debug off
SIP Debugging Disabled

AVAYA


10:12:32 SIP>INVITE sip:4970@asterisk.com SIP/2.0
10:12:32 SIP|From: sip:89144149111@invalid.unknown.domain;tag=0a8c46
10:12:32 SIP|322be7134b858b028b400
10:12:32 SIP|To: sip:4970@asterisk.com
10:12:32 SIP|Call-ID: 0a8c46322be7135b858b028b400
10:12:32 SIP|CSeq: 1 INVITE
10:12:32 SIP|Content-Length: 239
10:12:32     dial 4970 route:UDP|AAR
10:12:32     term trunk-group 70      cid 0x843
10:12:32     dial 4970 route:UDP|AAR
10:12:32     route-pattern  70 preference 1 location 1/ALL  cid 0x843
10:12:32     seize trunk-group 70 member 5    cid 0x843
10:12:32     Setup digits 4970
10:12:32     Calling Number & Name 89144149111 NO-CPName
10:12:32 SIP<sip 2.0="" 100="" trying="" <br="">
10:12:32 SIP|From: sip:89144149111@invalid.unknown.domain;Tag=0a8c46
10:12:32 SIP|322be7134b858b028b400
10:12:32 SIP|To: sip:4970@asterisk.com
10:12:32 SIP|Call-ID: 0a8c46322be7135b858b028b400
10:12:32 SIP|CSeq: 1 INVITE
10:12:32 SIP|Content-Length: 0
10:12:32     Proceed trunk-group 70 member 5    cid 0x843
10:12:32 SIP<sip 2.0="" 180="" ringing="" 10:12:32="" sip|from:="" sip:89144149111@invalid.unknown.domain;tag="0a8c46" 10:12:32="" sip|322be7134b858b028b400="" 10:12:32="" sip|to:="" sip:4970@asterisk.com;tag="as3e760790" 10:12:32="" sip|call-id:="" 0a8c46322be7135b858b028b400="" 10:12:32="" sip|cseq:="" 1="" invite="" 10:12:32="" sip|content-length:="" 0="" 10:12:32="" alert="" trunk-group="" 70="" member="" 5="" cid="" 0x843="" 10:12:32="" g729a="" ss:off="" ps:20="" rgn:1="" [10.214.129.204]:32720="" rgn:1="" [10.214.129.212]:2384="" 10:12:32="" xoip="" options:="" fax:t38="" modem:off="" tty:us="" uid:0x5007e="" xoip="" ip:="" [10.214.129.212]:2384="" 10:12:32="" sip<sip="" 2.0="" 180="" ringing="" 10:12:32="" sip|from:="" sip:89144149111@invalid.unknown.domain;tag="0a8c46" 10:12:32="" sip|322be7134b858b028b400="" 10:12:32="" sip|to:="" sip:4970@asterisk.com;tag="as3e760790" 10:12:32="" sip|call-id:="" 0a8c46322be7135b858b028b400="" 10:12:32="" sip|cseq:="" 1="" invite="" 10:12:32="" sip|content-length:="" 0="" 10:12:34="" sip<sip="" 2.0="" 200="" ok="" 10:12:34="" sip|from:="" sip:89144149111@invalid.unknown.domain;tag="0a8c46" 10:12:34="" sip|322be7134b858b028b400="" 10:12:34="" sip|to:="" sip:4970@asterisk.com;tag="as3e760790" 10:12:34="" sip|call-id:="" 0a8c46322be7135b858b028b400="" 10:12:34="" sip|cseq:="" 1="" invite="" 10:12:34="" sip|require:="" timer="" 10:12:34="" sip|content-length:="" 263="" 10:12:34="" sip="">ACK sip:4970@10.214.128.218:5060;transport=TCP SIP/2.0
10:12:34 SIP|From: sip:89144149111@invalid.unknown.domain;tag=0a8c46
10:12:34 SIP|322be7134b858b028b400
10:12:34 SIP|To: sip:4970@asterisk.com;tag=as3e760790
10:12:34 SIP|Call-ID: 0a8c46322be7135b858b028b400
10:12:34 SIP|CSeq: 1 ACK
10:12:34     active trunk-group 70 member 5    cid 0x843
10:12:34     G711A ss:off ps:20
             rgn:7 [10.214.128.218]:18056
             rgn:1 [10.214.129.212]:2396
10:12:34     xoip options: fax:Relay modem:off tty:US  uid:0x50073
             xoip ip: [10.214.129.212]:2396
10:12:34 SIP>BYE sip:4970@10.214.128.218:5060;transport=TCP SIP/2.0
10:12:34 SIP|From: sip:89144149111@invalid.unknown.domain;tag=0a8c46
10:12:34 SIP|322be7134b858b028b400
10:12:34 SIP|To: sip:4970@asterisk.com;tag=as3e760790
10:12:34 SIP|Call-ID: 0a8c46322be7135b858b028b400
10:12:34 SIP|CSeq: 2 BYE
10:12:34     idle trunk-group 45 member 1    cid 0x843

удалить закрыть спам изменить тег редактировать

спросил 2017-04-28 04:24:40 +0400

igkhv

1 ● 1 ● 1

обновил 2017-04-28 04:30:03 +0400

Comments

На asterisk - canreinvite=no стоит ? И версию с 11.2 наверно надо обновлять раз в 3 года ? И по логам - у вас NAT в локалке ? И отбой звонка прислала Avaya

awsswa ( 2017-04-28 09:54:39 +0400 )редактировать

canreinvite=no стоит. Версия стабильная и всем устраивает, АТСки одногодки. Между Астериксом и АТСкой NATa нет, только маршрутизируются две сетки. А из-за чего Авайка может присылать отбой? С городскими-то всё отлично...

igkhv ( 2017-05-02 03:16:58 +0400 )редактировать

У вас лог - с одним плечом - телефон и сервер asterisk. Нужно лог оба плеча - телефон-asterisk и asterisk-avaya. Не понятно с какой проблемой пришел отбой от avaya.

awsswa ( 2017-05-02 07:46:45 +0400 )редактировать

Знаете кого-то кто может ответить? Поделитесь ссылкой на этот вопрос через email.

Alcatel <-> Avaya <sip> FPBX-2.11 проблема с сотовыми телефонами

Comments

Будьте первым, кто ответит на этот вопрос!

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