проблема с прохождением звонков

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grenlan
Сообщения: 4
Зарегистрирован: 07 июл 2012, 16:26

проблема с прохождением звонков

Сообщение grenlan » 03 сен 2012, 12:55

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: Выделить всё

<--- Reliably Transmitting &#40;NAT&#41; to 82.215.234.207&#58;5062 --->
SIP/2.0 401 Unauthorized
Via&#58; SIP/2.0/UDP 82.215.234.207&#58;5062;branch=z9hG4bK-d8754z-2f5843f4ea7e8d2c-1---d8754z-;received=82.215.234.207;rport=5062
From&#58; "79124534221"<sip&#58;918149510793703@aster;transport=UDP>;tag=6178c336
To&#58; <sip&#58;79124534221@aster;transport=UDP>;tag=as3fa3e3c7
Call-ID&#58; NmVjYzlhYjAxOTMxN2Q5YzNlYTM5ZGIzYWU3NGY3YTM.
CSeq&#58; 1 INVITE
Server&#58; Asterisk PBX
Allow&#58; INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported&#58; replaces, timer
WWW-Authenticate&#58; Digest algorithm=MD5, realm="aster", nonce="1a01b7db"
Content-Length&#58; 0


<------------>
Scheduling destruction of SIP dialog 'NmVjYzlhYjAxOTMxN2Q5YzNlYTM5ZGIzYWU3NGY3YTM.' in 32000 ms &#40;Method&#58; INVITE&#41;

<--- SIP read from UDP&#58;82.215.234.207&#58;5062 --->
ACK sip&#58;79124534221@aster;transport=UDP SIP/2.0
Via&#58; SIP/2.0/UDP 82.215.234.207&#58;5062;branch=z9hG4bK-d8754z-2f5843f4ea7e8d2c-1---d8754z-
Max-Forwards&#58; 70
To&#58; <sip&#58;79124534221@aster;transport=UDP>;tag=as3fa3e3c7
From&#58; "79124534221"<sip&#58;918149510793703@aster;transport=UDP>;tag=6178c336
Call-ID&#58; NmVjYzlhYjAxOTMxN2Q5YzNlYTM5ZGIzYWU3NGY3YTM.
CSeq&#58; 1 ACK
Content-Length&#58; 0


<------------->
--- &#40;8 headers 0 lines&#41; ---

<--- SIP read from UDP&#58;82.215.234.207&#58;5062 --->
INVITE sip&#58;79124534221@aster;transport=UDP SIP/2.0
Via&#58; SIP/2.0/UDP 82.215.234.207&#58;5062;branch=z9hG4bK-d8754z-a628a1239cb24d94-1---d8754z-
Max-Forwards&#58; 70
Contact&#58; <sip&#58;918149510793703@82.215.234.207&#58;5062;transport=UDP>
To&#58; <sip&#58;79124534221@aster;transport=UDP>
From&#58; "79124534221"<sip&#58;918149510793703@aster;transport=UDP>;tag=6178c336
Call-ID&#58; NmVjYzlhYjAxOTMxN2Q5YzNlYTM5ZGIzYWU3NGY3YTM.
CSeq&#58; 2 INVITE
Allow&#58; INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type&#58; application/sdp
Supported&#58; replaces, norefersub, extended-refer, X-cisco-serviceuri
User-Agent&#58; Zoiper rev.11137
Authorization&#58; Digest username="918149510793703",realm="aster",nonce="1a01b7db",uri="sip&#58;79124534221@aster;transport=UDP",response="71b469d62ed3ed39e8e3d4a146ab6801",algorithm=MD5
Allow-Events&#58; presence, kpml
Content-Length&#58; 327

v=0
o=Zoiper_user 0 0 IN IP4 82.215.234.207
s=Zoiper_session
c--- &#40;15 headers 15 lines&#41; ---
Sending to 82.215.234.207&#58;5062 &#40;NAT&#41;
Using INVITE request as basis request - NmVjYzlhYjAxOTMxN2Q5YzNlYTM5ZGIzYWU3NGY3YTM.
Found peer '918149510793703' for '918149510793703' from 82.215.234.207&#58;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&#58; us - 0x10e &#40;gsm|ulaw|alaw|g729&#41;, peer - audio=0x60e &#40;gsm|ulaw|alaw|speex|ilbc&#41;/video=0x0 &#40;nothing&#41;/text=0x0 &#40;nothing&#41;, combined - 0xe &#40;gsm|ulaw|alaw&#41;
Non-codec capabilities &#40;dtmf&#41;&#58; us - 0x1 &#40;telephone-event|&#41;, peer - 0x1 &#40;telephone-event|&#41;, combined - 0x1 &#40;telephone-event|&#41;
Peer audio RTP is at port 82.215.234.207&#58;8000
Looking for 79124534221 in a2billing &#40;domain aster&#41;
list_route&#58; hop&#58; <sip&#58;918149510793703@82.215.234.207&#58;5062;transport=UDP>

<--- Transmitting &#40;NAT&#41; to 82.215.234.207&#58;5062 --->
SIP/2.0 100 Trying
Via&#58; SIP/2.0/UDP 82.215.234.207&#58;5062;branch=z9hG4bK-d8754z-a628a1239cb24d94-1---d8754z-;received=82.215.234.207;rport=5062
From&#58; "79124534221"<sip&#58;918149510793703@aster;transport=UDP>;tag=6178c336
To&#58; <sip&#58;79124534221@aster;transport=UDP>
Call-ID&#58; NmVjYzlhYjAxOTMxN2Q5YzNlYTM5ZGIzYWU3NGY3YTM.
CSeq&#58; 2 INVITE
Server&#58; Asterisk PBX
Allow&#58; INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported&#58; replaces, timer
Contact&#58; <sip&#58;79124534221@94.135.132.34&#58;5060>
Content-Length&#58; 0


<------------>
Audio is at 14992
Adding codec 0x4 &#40;ulaw&#41; to SDP
Adding codec 0x8 &#40;alaw&#41; to SDP
Adding codec 0x2 &#40;gsm&#41; to SDP
Adding non-codec 0x1 &#40;telephone-event&#41; to SDP

<--- Reliably Transmitting &#40;NAT&#41; to 82.215.234.207&#58;5062 --->
SIP/2.0 200 OK
Via&#58; SIP/2.0/UDP 82.215.234.207&#58;5062;branch=z9hG4bK-d8754z-a628a1239cb24d94-1---d8754z-;received=82.215.234.207;rport=5062
From&#58; "79124534221"<sip&#58;918149510793703@aster;transport=UDP>;tag=6178c336
To&#58; <sip&#58;79124534221@aster;transport=UDP>;tag=as604af1c7
Call-ID&#58; NmVjYzlhYjAxOTMxN2Q5YzNlYTM5ZGIzYWU3NGY3YTM.
CSeq&#58; 2 INVITE
Server&#58; Asterisk PBX
Allow&#58; INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported&#58; replaces, timer
Contact&#58; <sip&#58;79124534221@94.135.132.34&#58;5060>
Content-Type&#58; application/sdp
Content-Length&#58; 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&#58;0 PCMU/8000
a=rtpmap&#58;8 PCMA/8000
a=rtpmap&#58;3 GSM/8000
a=rtpmap&#58;101 telephone-event/8000
a=fmtp&#58;101 0-16
a=ptime&#58;20
a=sendrecv

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

<--- SIP read from UDP&#58;82.215.234.207&#58;5062 --->
ACK sip&#58;79124534221@94.135.132.34&#58;5060 SIP/2.0
Via&#58; SIP/2.0/UDP 82.215.234.207&#58;5062;branch=z9hG4bK-d8754z-792c5e932fdec75e-1---d8754z-
Max-Forwards&#58; 70
Contact&#58; <sip&#58;918149510793703@82.215.234.207&#58;5062;transport=UDP>
To&#58; <sip&#58;79124534221@aster;transport=UDP>;tag=as604af1c7
From&#58; "79124534221"<sip&#58;918149510793703@aster;transport=UDP>;tag=6178c336
Call-ID&#58; NmVjYzlhYjAxOTMxN2Q5YzNlYTM5ZGIzYWU3NGY3YTM.
CSeq&#58; 2 ACK
User-Agent&#58; Zoiper rev.11137
Authorization&#58; Digest username="918149510793703",realm="aster",nonce="1a01b7db",uri="sip&#58;79124534221@aster;transport=UDP",response="71b469d62ed3ed39e8e3d4a146ab6801",algorithm=MD5
Content-Length&#58; 0


<------------->
--- &#40;11 headers 0 lines&#41; ---
Scheduling destruction of SIP dialog 'NmVjYzlhYjAxOTMxN2Q5YzNlYTM5ZGIzYWU3NGY3YTM.' in 32000 ms &#40;Method&#58; ACK&#41;
set_destination&#58; Parsing <sip&#58;918149510793703@82.215.234.207&#58;5062;transport=UDP> for address/port to send to
set_destination&#58; set destination to 82.215.234.207&#58;5062
Reliably Transmitting &#40;NAT&#41; to 82.215.234.207&#58;5062&#58;
BYE sip&#58;918149510793703@82.215.234.207&#58;5062;transport=UDP SIP/2.0
Via&#58; SIP/2.0/UDP 94.135.132.34&#58;5060;branch=z9hG4bK527df51e;rport
Max-Forwards&#58; 70
From&#58; <sip&#58;79124534221@aster;transport=UDP>;tag=as604af1c7
To&#58; "79124534221"<sip&#58;918149510793703@aster;transport=UDP>;tag=6178c336
Call-ID&#58; NmVjYzlhYjAxOTMxN2Q5YzNlYTM5ZGIzYWU3NGY3YTM.
CSeq&#58; 102 BYE
User-Agent&#58; Asterisk PBX
Proxy-Authorization&#58; Digest username="918149510793703", realm="aster", algorithm=MD5, uri="sip&#58;aster", nonce="", response="f434248b20df292989f2c3051af263de"
X-Asterisk-HangupCause&#58; Normal Clearing
X-Asterisk-HangupCauseCode&#58; 16
Content-Length&#58; 0


---

<--- SIP read from UDP&#58;82.215.234.207&#58;5062 --->
SIP/2.0 200 OK
Via&#58; SIP/2.0/UDP 94.135.132.34&#58;5060;branch=z9hG4bK527df51e;rport=5060
Contact&#58; <sip&#58;918149510793703@82.215.234.207&#58;5062;transport=UDP>
To&#58; "79124534221"<sip&#58;918149510793703@aster;transport=UDP>;tag=6178c336
From&#58; <sip&#58;79124534221@aster;transport=UDP>;tag=as604af1c7
Call-ID&#58; NmVjYzlhYjAxOTMxN2Q5YzNlYTM5ZGIzYWU3NGY3YTM.
CSeq&#58; 102 BYE
User-Agent&#58; Zoiper rev.11137
Content-Length&#58; 0


<------------->
--- &#40;9 headers 0 lines&#41; ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog 'NmVjYzlhYjAxOTMxN2Q5YzNlYTM5ZGIzYWU3NGY3YTM.' Method&#58; ACK
Может кто нибудь сможет подсказать что ему не нравится?

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