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Откуда: Sakhalin
Сообщений: 14
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FXO-Cisco->SIP-Trixbox прошу помочь
Добрый день
Имееться Cisco с FXO на циске:
voice-port 2/0/16
supervisory disconnect dualtone mid-call
supervisory custom-cptone mm
supervisory dualtone-detect-params 1
disc_pi_off
input gain 5
timeouts call-disconnect 5
timeouts wait-release 5
connection plar 422044
station-id number 0422044
caller-id enable
dial-peer voice 422044 voip
service session
destination-pattern 422044
voice-class codec 1
session protocol sipv2
session target sip-server
session transport udp
dtmf-relay rtp-nte
fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback cisco
no vad
!
sip-ua
credentials username asterisk password xxxxxx realm trixbox.gildia.lan
authentication username asterisk password xxxxxxxx
retry invite 3
retry response 3
retry bye 3
retry cancel 3
retry register 10
retry subscribe 10
timers trying 1000
timers register 150
registrar dns:trixbox.gildia.lan expires 3600
sip-server dns:trixbox.gildia.lan
Звонок удачно приходит до Asteriska
там он попадаете в транк:
Исходящие настройки
опции для PEER
username=asterisk
type=peer
secret=passwd
qualify=yes
nat=no
insecure=invite,port
language=ru
host=192.168.0.242
fromuser=asterisk
dTMFMode=rfc2833
disallow=all
canreinvite=no
callerid=422044
auth=md5,plaintext
allow=g711&ulaw&alaw
Установки для входящих соединений
Контекст USER c2821
username=asterisk
type=peer
secret=passwd
qualify=yes
nat=no
insecure=invite,port
language=ru
host=192.168.0.242
dTMFMode=rfc2833
disallow=all
canreinvite=no
callerid=422044
auth=md5,plaintext
allow=g711&ulaw&alaw
Во входящих маршрутах настроено сл.
входящий с DID 422044
направляется в группу вызовов X.
На астериске видно что cisco вроде как зарегистрировалась...
trixbox*CLI> sip show peers
Name/username Host Dyn Nat ACL Port Status
c2821.gildia.lan/asterisk 192.168.0.242 5060 OK (6 ms)
c2821/asterisk 192.168.0.242 5060 OK (6 ms)
при разрешении анонимных звонков всё отлично работает но анонимные звонки не гуд...
в логах вижу следующее
--- (17 headers 11 lines) ---
Really destroying SIP dialog '6ad028ca4b8abab442487c6b08b4e55b@192.168.0.235' Method: OPTIONS
-- Remote UNIX connection
-- Remote UNIX connection disconnected
-- Remote UNIX connection
-- Remote UNIX connection disconnected
-- Remote UNIX connection
-- Remote UNIX connection disconnected
Really destroying SIP dialog 'D1B9-76D9-468395479B11B685B423-264@SipHost' Method: REGISTER
-- Remote UNIX connection
-- Remote UNIX connection disconnected
trixbox*CLI>
<--- SIP read from UDP://192.168.0.242:49649 --->
INVITE sip:422044@trixbox.gildia.lan:5060 SIP/2.0
Date: Wed, 21 Oct 2009 02:50:46 GMT
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
From: <sip:0422044@trixbox.gildia.lan>;tag=4BFEA54-2434
Allow-Events: telephone-event
Supported: 100rel,timer,resource-priority,replaces,histinfo,sdp-anat
Min-SE: 1800
Remote-Party-ID: <sip:0422044@192.168.0.242>;party=calling;screen=no;privacy=off
Cisco-Guid: 1576631794-3173192158-2432362545-396813358
Timestamp: 1256093446
Content-Length: 357
User-Agent: Cisco-SIPGateway/IOS-12.x
To: <sip:422044@trixbox.gildia.lan>
Contact: <sip:0422044@192.168.0.242:5060>
Expires: 180
Content-Disposition: session;handling=required
Content-Type: application/sdp
Call-ID: 5DFB526A-BD2311DE-90FFE831-17A6E42E@192.168.0.242
History-Info: <sip:422044@trixbox.gildia.lan:5060>;index=1
Via: SIP/2.0/UDP 192.168.0.242:5060;branch=z9hG4bK235016D3
CSeq: 101 INVITE
Max-Forwards: 70
v=0
o=CiscoSystemsSIP-GW-UserAgent 1625 251 IN IP4 192.168.0.242
s=SIP Call
c=IN IP4 192.168.0.242
t=0 0
m=audio 17554 RTP/AVP 8 0 18 100 101
c=IN IP4 192.168.0.242
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:100 X-NSE/8000
a=fmtp:100 192-194
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
<------------->
--- (22 headers 15 lines) ---
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Using SIP VRTP TOS bits 136
== Using SIP VRTP CoS mark 6
Sending to 192.168.0.242 : 5060 (no NAT)
Using INVITE request as basis request - 5DFB526A-BD2311DE-90FFE831-17A6E42E@192.168.0.242
No user '0422044' in SIP users list
Found peer 'c2821' for '0422044' from 192.168.0.242:49649
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 100
Found RTP audio format 101
Peer audio RTP is at port 192.168.0.242:17554
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format G729 for ID 18
Found unknown media description format X-NSE for ID 100
Found audio description format telephone-event for ID 101
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.0.242:17554
Looking for 422044 in from-sip-external (domain trixbox.gildia.lan)
list_route: hop: <sip:0422044@192.168.0.242:5060>
trixbox*CLI>
<--- Transmitting (no NAT) to 192.168.0.242:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.242:5060;branch=z9hG4bK235016D3;received=192.168.0.242
From: <sip:0422044@trixbox.gildia.lan>;tag=4BFEA54-2434
To: <sip:422044@trixbox.gildia.lan>
Call-ID: 5DFB526A-BD2311DE-90FFE831-17A6E42E@192.168.0.242
CSeq: 101 INVITE
User-Agent: Asterisk PBX 1.6.0.9-samy-r27
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:422044@192.168.0.235>
Content-Length: 0
<------------>
-- Executing [422044@from-sip-external:1] NoOp("SIP/asterisk-b783c7e0", "Received incoming SIP connection from unknown peer to 422044") in new stack
-- Executing [422044@from-sip-external:2] Set("SIP/asterisk-b783c7e0", "DID=422044") in new stack
-- Executing [422044@from-sip-external:3] Goto("SIP/asterisk-b783c7e0", "s,1") in new stack
-- Goto (from-sip-external,s,1)
-- Executing [s@from-sip-external:1] GotoIf("SIP/asterisk-b783c7e0", "0?from-trunk,422044,1") in new stack
-- Executing [s@from-sip-external:2] Set("SIP/asterisk-b783c7e0", "TIMEOUT(absolute)=15") in new stack
Channel will hangup at 2009-10-21 13:45:38.000 VLAST.
-- Executing [s@from-sip-external:3] Answer("SIP/asterisk-b783c7e0", "") in new stack
Audio is at 192.168.0.235 port 18188
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Reliably Transmitting (no NAT) to 192.168.0.242:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.242:5060;branch=z9hG4bK235016D3;received=192.168.0.242
From: <sip:0422044@trixbox.gildia.lan>;tag=4BFEA54-2434
To: <sip:422044@trixbox.gildia.lan>;tag=as64df722e
Call-ID: 5DFB526A-BD2311DE-90FFE831-17A6E42E@192.168.0.242
CSeq: 101 INVITE
User-Agent: Asterisk PBX 1.6.0.9-samy-r27
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:422044@192.168.0.235>
Content-Type: application/sdp
Content-Length: 298
Интересует вот эта строчка я так понимаю в ней вся проблема
Executing [422044@from-sip-external:1] NoOp("SIP/asterisk-b783c7e0", "Received incoming SIP connection from unknown peer to 422044") in new stack
почему peer неизвестный если выше явно сказано что соответствие найдено
Found peer 'c2821' for '0422044' from 192.168.0.242:49649.
Прошу помочь разобраться, почему не доходят звонки в группу.
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