Не работает исходящий SIP звонок. Не знаю куда копать.


trixbox1*CLI> sip show peers
Name/username Host Dyn Nat ACL Port Status
internetcalls/skynet66 77.72.169.129 5060 Unmonitored
gts-sip/5620640 89.232.125.48 N 5060 Unmonitored
105 (Unspecified) D N A 5060 UNKNOWN
104 (Unspecified) D N A 5060 UNKNOWN
103/103 192.168.101.154 D N A 5061 OK (16 ms)
102/102 192.168.101.154 D N A 5060 OK (16 ms)
101/101 192.168.101.153 D N A 5072 OK (15 ms)
100/100 192.168.101.153 D N A 5071 OK (15 ms)
8 sip peers [Monitored: 4 online, 2 offline Unmonitored: 2 online, 0 offline]

trixbox1*CLI> sip show registry
Host Username Refresh State Reg.Time
89.xxx.125.48:5060 5620640 101 Registered Fri, 19 Mar 2010 12:10:51
1 SIP registrations.
Really destroying SIP dialog '11ce763328da47402307a074021634c5@127.0.0.1' Method: REGISTER



PEER Details:
host=89.232.125.48
insecure=port,invite
nat=yes
fromuser=5620640
username=5620640
secret=xxxxxxxx
type=peer
disallow=all
allow=ulaw&g711a&g711


а это кусочек лога с SIP/2.0 403 от SIP прокси:

-- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/103-08d2bf20", "0?customtrunk") in new stack
-- Executing [s@macro-dialout-trunk:19] Dial("SIP/103-08d2bf20", "SIP/gts-sip/2909294,300,") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Using SIP VRTP TOS bits 136
== Using SIP VRTP CoS mark 6
Audio is at 192.168.101.180 port 16400
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 89.xxx.125.48:5060:
INVITE sip:2909294@89.xxx.125.48 SIP/2.0
Via: SIP/2.0/UDP 192.168.101.180:5060;branch=z9hG4bK0c37c7a5;rport
Max-Forwards: 70
From: "103" <sip:5620640@192.168.101.180>;tag=as4d695e92
To: <sip:2909294@89.xxx.125.48>
Contact: <sip:5620640@192.168.101.180>
Call-ID: 75408280436ca6a31df57c0f30f3e8d6@192.168.101.180
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.0.9-samy-r27
ate: Fri, 19 Mar 2010 09:02:49 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 278

v=0
o=root 1153224551 1153224551 IN IP4 192.168.101.180
s=Asterisk PBX 1.6.0.9-samy-r27
c=IN IP4 192.168.101.180
t=0 0
m=audio 16400 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
-- Called gts-sip/2909294
trixbox1*CLI>
<--- SIP read from UDP://89.xxx.125.48:5060 --->
SIP/2.0 100 Trying
From: "103"<sip:5620640@192.168.101.180>;tag=as4d695e92
To: <sip:2909294@89.xxx.125.48>
Call-ID: 75408280436ca6a31df57c0f30f3e8d6@192.168.101.180
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.101.180:5060;received=78.138.144.127;rport=5060;branch=z9hG4bK0c37c7a5
Content-Length: 0


<------------->
--- (7 headers 0 lines) ---
trixbox1*CLI>
<--- SIP read from UDP://89.xxx.125.48:5060 --->
SIP/2.0 403 Forbidden
From: "103"<sip:5620640@192.168.101.180>;tag=as4d695e92
To: <sip:2909294@89.xxx.125.48>;tag=2083351771
Call-ID: 75408280436ca6a31df57c0f30f3e8d6@192.168.101.180
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.101.180:5060;received=78.138.144.127;rport=5060;branch=z9hG4bK0c37c7a5
contact: <sip:2909294@tattele.com:5060;maddr=89.xxx.125.48>
supported: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec,join,com.nortelnetworks.im.encryption
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---
Transmitting (NAT) to 89.xx.125.48:5060:
ACK sip:2909294@89.xxx.125.48 SIP/2.0
Via: SIP/2.0/UDP 192.168.101.180:5060;branch=z9hG4bK0c37c7a5;rport
Max-Forwards: 70
From: "103" <sip:5620640@192.168.101.180>;tag=as4d695e92
To: <sip:2909294@89.xxx.125.48>;tag=2083351771
Contact: <sip:5620640@192.168.101.180>
Call-ID: 75408280436ca6a31df57c0f30f3e8d6@192.168.101.180
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.0.9-samy-r27
Content-Length: 0


---
-- SIP/gts-sip-b79a2530 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing [s@macro-dialout-trunk:20] Goto("SIP/103-08d2bf20", "s-CONGESTION,1") in new stack
-- Goto (macro-dialout-trunk,s-CONGESTION,1)
-- Executing [s-CONGESTION@macro-dialout-trunk:1] GotoIf("SIP/103-08d2bf20", "1?noreport") in new stack
-- Goto (macro-dialout-trunk,s-CONGESTION,3)
-- Executing [s-CONGESTION@macro-dialout-trunk:3] NoOp("SIP/103-08d2bf20", "TRUNK Dial failed due to CONGESTION - failing through to other trunks") in new stack
-- Executing [78432909294@from-internal:5] Macro("SIP/103-08d2bf20", "outisbusy,") in new stack
-- Executing [s@macro-outisbusy:1] Playback("SIP/103-08d2bf20", "all-circuits-busy-now,noanswer") in new stack
-- <SIP/103-08d2bf20> Playing 'all-circuits-busy-now.ulaw' (language 'en')
Really destroying SIP dialog '75408280436ca6a31df57c0f30f3e8d6@192.168.101.180' Method: INVITE
-- Executing [s@macro-outisbusy:2] Playback("SIP/103-08d2bf20", "pls-try-call-later,noanswer") in new stack
-- <SIP/103-08d2bf20> Playing 'pls-try-call-later.ulaw' (language 'en')
-- Executing [s@macro-outisbusy:3] Macro("SIP/103-08d2bf20", "hangupcall") in new stack
-- Executing [s@macro-hangupcall:1] ResetCDR("SIP/103-08d2bf20", "vw") in new stack
-- Executing [s@macro-hangupcall:2] NoCDR("SIP/103-08d2bf20", "") in new stack
-- Executing [s@macro-hangupcall:3] GotoIf("SIP/103-08d2bf20", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,6)
-- Executing [s@macro-hangupcall:6] GotoIf("SIP/103-08d2bf20", "1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [s@macro-hangupcall:9] GotoIf("SIP/103-08d2bf20", "1?theend") in new stack
-- Goto (macro-hangupcall,s,11)
-- Executing [s@macro-hangupcall:11] Hangup("SIP/103-08d2bf20", "") in new stack
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/103-08d2bf20' in macro 'hangupcall'
== Spawn extension (macro-outisbusy, s, 3) exited non-zero on 'SIP/103-08d2bf20' in macro 'outisbusy'
== Spawn extension (from-internal, 78432909294, 5) exited non-zero on 'SIP/103-08d2bf20'
-- Executing [h@from-internal:1] Macro("SIP/103-08d2bf20", "hangupcall") in new stack
-- Executing [s@macro-hangupcall:1] ResetCDR("SIP/103-08d2bf20", "vw") in new stack
-- Executing [s@macro-hangupcall:2] NoCDR("SIP/103-08d2bf20", "") in new stack
-- Executing [s@macro-hangupcall:3] GotoIf("SIP/103-08d2bf20", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,6)
-- Executing [s@macro-hangupcall:6] GotoIf("SIP/103-08d2bf20", "1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [s@macro-hangupcall:9] GotoIf("SIP/103-08d2bf20", "1?theend") in new stack
-- Goto (macro-hangupcall,s,11)
-- Executing [s@macro-hangupcall:11] Hangup("SIP/103-08d2bf20", "") in new stack
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/103-08d2bf20' in macro 'hangupcall'
== Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/103-08d2bf20'
== End MixMonitor Recording SIP/103-08d2bf20

SIP/2.0 403 Forbidden говорит. Нельзя, говорит, звонить не зарегистрировавшись.
Согласно регистрации Астериск видит, что 103 на адресе 192.168.101.154
103/103 192.168.101.154
А звонок идёт якобы от 103 совсем с другого адреса.
From: "103" <sip:5620640@192.168.101.180>;

101.180 - адрес сервера астерикс
101.154 - адрес сип трубки (локальный экстеншн) с которого номер набираем.

Не подскажете в каком направлении копать?

Насколько я понял по списку регистраций мы зарегистрированы на сервере сип:

trixbox1*CLI> sip show registry
Host Username Refresh State Reg.Time
89.xxx.125.48:5060 5620640 101 Registered Fri, 19 Mar 2010 12:10:51
1 SIP registrations.
Really destroying SIP dialog '11ce763328da47402307a074021634c5@127.0.0.1' Method: REGISTER

allow=ulaw&g711a&g711 - что это?



зачем для пира nat=yes ?
и неужто ГТС не может вам помочь с настройкой?

Вы правы:) Исправил, ребутнул, результат тот же. Новый PEER details:

host=89.232.125.48
insecure=port,invite
fromuser=5620640
username=5620640
secret=VPMYp1AF12
type=peer
disallow=all
allow=ulaw&alaw

ГТС могут помочь только с настройкой железки, но не астерикс, а X-Lite подцепленный на этом же IP с виндовой машины работает на ура с этим SIP прокси. Т.е. дело в trixbox.

не знаю с виду всё правильно , может им fromip прописать , но слабо вериться .
а номер 2909294, что Вы набрали - правильный? попробуйте с кодом города , без кода сттаны , но с кодом города. а то мало ли у них диалплан напутан.
Входящие работают?

2909294 правильный.. И набирается отовсюду легко. И с кодом города такая же засада. SIP 2.0 / 403 Может у меня диалплан напутан? Что сюда запостить чтобы проверить?

dinskuns: Иправьте свое сообщение и удалите креды

попробуйте добавить
fromdomain=89.232.125.48

Возможно и диалплан не пропускает.
Проверить - создать
exten => _X.,1,Dial(SIP/TATtelecom/${EXTEN})
и побросать туда звонки.