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проблема с прохождением звонков

 
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grenlan



Зарегистрирован:
07.07.2012
Сообщения: 4

Статус: Оффлайн 

СообщениеДобавлено: Пн Сен 03, 2012 12:55    Заголовок сообщения: проблема с прохождением звонков

Добрый день. Решил начать изучать Asterisk, настроил по мануалам связку a2billing-a c asterisk-ом, попробывал создать пользователя и зарегистрироватся, все удалось но при попытке позвонить, через 2 секунды соединение сбрасывается в логах при этом можно видеть следующее:
Код:
<--- Reliably Transmitting (NAT) to 82.215.234.207:5062 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 82.215.234.207:5062;branch=z9hG4bK-d8754z-2f5843f4ea7e8d2c-1---d8754z-;received=82.215.234.207;rport=5062
From: "79124534221"<sip:918149510793703@aster;transport=UDP>;tag=6178c336
To: <sip:79124534221@aster;transport=UDP>;tag=as3fa3e3c7
Call-ID: NmVjYzlhYjAxOTMxN2Q5YzNlYTM5ZGIzYWU3NGY3YTM.
CSeq: 1 INVITE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="aster", nonce="1a01b7db"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'NmVjYzlhYjAxOTMxN2Q5YzNlYTM5ZGIzYWU3NGY3YTM.' in 32000 ms (Method: INVITE)

<--- SIP read from UDP:82.215.234.207:5062 --->
ACK sip:79124534221@aster;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 82.215.234.207:5062;branch=z9hG4bK-d8754z-2f5843f4ea7e8d2c-1---d8754z-
Max-Forwards: 70
To: <sip:79124534221@aster;transport=UDP>;tag=as3fa3e3c7
From: "79124534221"<sip:918149510793703@aster;transport=UDP>;tag=6178c336
Call-ID: NmVjYzlhYjAxOTMxN2Q5YzNlYTM5ZGIzYWU3NGY3YTM.
CSeq: 1 ACK
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---

<--- SIP read from UDP:82.215.234.207:5062 --->
INVITE sip:79124534221@aster;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 82.215.234.207:5062;branch=z9hG4bK-d8754z-a628a1239cb24d94-1---d8754z-
Max-Forwards: 70
Contact: <sip:918149510793703@82.215.234.207:5062;transport=UDP>
To: <sip:79124534221@aster;transport=UDP>
From: "79124534221"<sip:918149510793703@aster;transport=UDP>;tag=6178c336
Call-ID: NmVjYzlhYjAxOTMxN2Q5YzNlYTM5ZGIzYWU3NGY3YTM.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
Supported: replaces, norefersub, extended-refer, X-cisco-serviceuri
User-Agent: Zoiper rev.11137
Authorization: Digest username="918149510793703",realm="aster",nonce="1a01b7db",uri="sip:79124534221@aster;transport=UDP",response="71b469d62ed3ed39e8e3d4a146ab6801",algorithm=MD5
Allow-Events: presence, kpml
Content-Length: 327

v=0
o=Zoiper_user 0 0 IN IP4 82.215.234.207
s=Zoiper_session
c--- (15 headers 15 lines) ---
Sending to 82.215.234.207:5062 (NAT)
Using INVITE request as basis request - NmVjYzlhYjAxOTMxN2Q5YzNlYTM5ZGIzYWU3NGY3YTM.
Found peer '918149510793703' for '918149510793703' from 82.215.234.207:5062
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 110
Found RTP audio format 98
Found RTP audio format 101
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format speex for ID 110
Found audio description format iLBC for ID 98
Found audio description format telephone-event for ID 101
Capabilities: us - 0x10e (gsm|ulaw|alaw|g729), peer - audio=0x60e (gsm|ulaw|alaw|speex|ilbc)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 82.215.234.207:8000
Looking for 79124534221 in a2billing (domain aster)
list_route: hop: <sip:918149510793703@82.215.234.207:5062;transport=UDP>

<--- Transmitting (NAT) to 82.215.234.207:5062 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 82.215.234.207:5062;branch=z9hG4bK-d8754z-a628a1239cb24d94-1---d8754z-;received=82.215.234.207;rport=5062
From: "79124534221"<sip:918149510793703@aster;transport=UDP>;tag=6178c336
To: <sip:79124534221@aster;transport=UDP>
Call-ID: NmVjYzlhYjAxOTMxN2Q5YzNlYTM5ZGIzYWU3NGY3YTM.
CSeq: 2 INVITE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:79124534221@94.135.132.34:5060>
Content-Length: 0


<------------>
Audio is at 14992
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (NAT) to 82.215.234.207:5062 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 82.215.234.207:5062;branch=z9hG4bK-d8754z-a628a1239cb24d94-1---d8754z-;received=82.215.234.207;rport=5062
From: "79124534221"<sip:918149510793703@aster;transport=UDP>;tag=6178c336
To: <sip:79124534221@aster;transport=UDP>;tag=as604af1c7
Call-ID: NmVjYzlhYjAxOTMxN2Q5YzNlYTM5ZGIzYWU3NGY3YTM.
CSeq: 2 INVITE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:79124534221@94.135.132.34:5060>
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 1754016360 1754016360 IN IP4 94.135.132.34
s=Asterisk PBX 1.8.13.0
c=IN IP4 94.135.132.34
t=0 0
m=audio 14992 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:82.215.234.207:5062 --->
ACK sip:79124534221@94.135.132.34:5060 SIP/2.0
Via: SIP/2.0/UDP 82.215.234.207:5062;branch=z9hG4bK-d8754z-792c5e932fdec75e-1---d8754z-
Max-Forwards: 70
Contact: <sip:918149510793703@82.215.234.207:5062;transport=UDP>
To: <sip:79124534221@aster;transport=UDP>;tag=as604af1c7
From: "79124534221"<sip:918149510793703@aster;transport=UDP>;tag=6178c336
Call-ID: NmVjYzlhYjAxOTMxN2Q5YzNlYTM5ZGIzYWU3NGY3YTM.
CSeq: 2 ACK
User-Agent: Zoiper rev.11137
Authorization: Digest username="918149510793703",realm="aster",nonce="1a01b7db",uri="sip:79124534221@aster;transport=UDP",response="71b469d62ed3ed39e8e3d4a146ab6801",algorithm=MD5
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---
Scheduling destruction of SIP dialog 'NmVjYzlhYjAxOTMxN2Q5YzNlYTM5ZGIzYWU3NGY3YTM.' in 32000 ms (Method: ACK)
set_destination: Parsing <sip:918149510793703@82.215.234.207:5062;transport=UDP> for address/port to send to
set_destination: set destination to 82.215.234.207:5062
Reliably Transmitting (NAT) to 82.215.234.207:5062:
BYE sip:918149510793703@82.215.234.207:5062;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 94.135.132.34:5060;branch=z9hG4bK527df51e;rport
Max-Forwards: 70
From: <sip:79124534221@aster;transport=UDP>;tag=as604af1c7
To: "79124534221"<sip:918149510793703@aster;transport=UDP>;tag=6178c336
Call-ID: NmVjYzlhYjAxOTMxN2Q5YzNlYTM5ZGIzYWU3NGY3YTM.
CSeq: 102 BYE
User-Agent: Asterisk PBX
Proxy-Authorization: Digest username="918149510793703", realm="aster", algorithm=MD5, uri="sip:aster", nonce="", response="f434248b20df292989f2c3051af263de"
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---

<--- SIP read from UDP:82.215.234.207:5062 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 94.135.132.34:5060;branch=z9hG4bK527df51e;rport=5060
Contact: <sip:918149510793703@82.215.234.207:5062;transport=UDP>
To: "79124534221"<sip:918149510793703@aster;transport=UDP>;tag=6178c336
From: <sip:79124534221@aster;transport=UDP>;tag=as604af1c7
Call-ID: NmVjYzlhYjAxOTMxN2Q5YzNlYTM5ZGIzYWU3NGY3YTM.
CSeq: 102 BYE
User-Agent: Zoiper rev.11137
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog 'NmVjYzlhYjAxOTMxN2Q5YzNlYTM5ZGIzYWU3NGY3YTM.' Method: ACK

Может кто нибудь сможет подсказать что ему не нравится?
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