проблема с прохождением звонков
| Код: |
| 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";tag=6178c336 To: ;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) 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: ;tag=as3fa3e3c7 From: "79124534221";tag=6178c336 Call-ID: NmVjYzlhYjAxOTMxN2Q5YzNlYTM5ZGIzYWU3NGY3YTM. CSeq: 1 ACK Content-Length: 0 --- (8 headers 0 lines) --- 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: To: From: "79124534221";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/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";tag=6178c336 To: Call-ID: NmVjYzlhYjAxOTMxN2Q5YzNlYTM5ZGIzYWU3NGY3YTM. CSeq: 2 INVITE Server: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: 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 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";tag=6178c336 To: ;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: 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 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: To: ;tag=as604af1c7 From: "79124534221";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 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: ;tag=as604af1c7 To: "79124534221";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/2.0 200 OK Via: SIP/2.0/UDP 94.135.132.34:5060;branch=z9hG4bK527df51e;rport=5060 Contact: To: "79124534221";tag=6178c336 From: ;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?