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DVG-7062S пропадает pptp связь при SIP вызове

Модераторы: Victor Kolosov, Sergei Asmankin, Sergik

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Сообщение

Alexey Motkov

Заголовок сообщения: Re: DVG-7062S пропадает pptp связь при SIP вызове

Добавлено: Чт апр 21, 2011 10:32

Сотрудник D-LINK

Зарегистрирован: Пн янв 11, 2010 09:40
Сообщений: 4400

Disconnect потому, что пришел отбой от сервера. А лог входящего звонка можете выложить.

Вернуться наверх

Maksim237

Заголовок сообщения: Re: DVG-7062S пропадает pptp связь при SIP вызове

Добавлено: Пт апр 22, 2011 01:37

Зарегистрирован: Пт дек 10, 2010 03:42
Сообщений: 47
Откуда: Хабаровск

где вы видите отбой от сервера?

Код:

16:08 [562115] <---- 192.168.1.45:5060
16:08 [562115] INVITE sip:218@192.168.255.103 SIP/2.0
16:08 [562115] Via: SIP/2.0/UDP 192.168.1.45:5060;branch=z9hG4bK75ace634;rport
16:08 [562115] From: "0140" <sip:mirs_vs_vld_plsk_2@192.168.1.45>;tag=as486047ec
16:08 [562115] To: <sip:218@192.168.255.103>
16:08 [562115] Contact: <sip:mirs_vs_vld_plsk_2@192.168.1.45>
16:08 [562115] Call-ID: 65580ace4f6618c75077c9d9282531a8@192.168.1.45
16:08 [562115] CSeq: 102 INVITE
16:08 [562115] User-Agent: Asterisk PBX
16:08 [562115] Max-Forwards: 70
16:08 [562115] Date: Thu, 21 Apr 2011 22:16:05 GMT
16:08 [562115] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
16:08 [562115] Supported: replaces
16:08 [562115] Content-Type: application/sdp
16:08 [562115] Content-Length: 240
16:08 [562115] v=0
16:08 [562115] o=root 22539 22539 IN IP4 192.168.1.45
16:08 [562115] s=session
16:08 [562115] c=IN IP4 192.168.1.45
16:08 [562115] t=0 0
16:09 [562115] m=audio 17362 RTP/AVP 8 101
16:09 [562115] a=rtpmap:8 PCMA/8000
16:09 [562115] a=rtpmap:101 telephone-event/8000
16:09 [562115] a=fmtp:101 0-16
16:09 [562115] a=silenceSupp:off - - - -
16:09 [562115] a=ptime:20
16:09 [562115] a=sendrecv
16:09 [562115] -----
16:09 [562116] ----> 192.168.1.45:5060
16:09 [562116] SIP/2.0 100 Trying
16:09 [562116] Via:SIP/2.0/UDP 192.168.1.45:5060;rport;branch=z9hG4bK75ace634
16:09 [562116] From: "0140" <sip:mirs_vs_vld_plsk_2@192.168.1.45>;tag=as486047ec
16:09 [562116] To: <sip:218@192.168.255.103>
16:09 [562116] Call-ID:65580ace4f6618c75077c9d9282531a8@192.168.1.45
16:09 [562116] CSeq:102 INVITE
16:09 [562116] Content-Type:application/sdp
16:09 [562116] Content-Length:0
16:09 [562116] -----
16:09 [562117] 0: 218=OFFERING
16:09 [562117] 0: Get CallerId=0140
16:09 [562117] 0: Peer=192.168.1.45:17362, PT=8, RecvOnly=0
16:09 [562117] 0: ==12:Alerting
16:09 [562117] ----> 192.168.1.45:5060
16:09 [562118] SIP/2.0 180 Ringing
16:09 [562118] Via:SIP/2.0/UDP 192.168.1.45:5060;rport;branch=z9hG4bK75ace634
16:09 [562118] From: "0140" <sip:mirs_vs_vld_plsk_2@192.168.1.45>;tag=as486047ec
16:09 [562118] To: <sip:218@192.168.255.103>;tag=bb2963b0-741023
16:09 [562118] Call-ID:65580ace4f6618c75077c9d9282531a8@192.168.1.45
16:09 [562118] CSeq:102 INVITE
16:09 [562118] Contact:<sip:218@192.168.255.103:5060>
16:09 [562118] User-Agent:dlink 12-38-28928749-0.9.5.1.1140-IAD20
16:09 [562118] Content-Length:0
16:09 [562118] -----
16:17 [562202] 0: Fxs Pickup
16:17 [562203] 0: 218=ACCEPT
16:17 [562203] 0: ==13:VoipAnswering
16:17 [562203] ----> 192.168.1.45:5060
16:17 [562203] SIP/2.0 200 OK
16:17 [562203] Allow:INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,REFER,SUBSCRIBE,NOTIFY,UPDATE
16:17 [562203] Via:SIP/2.0/UDP 192.168.1.45:5060;rport;branch=z9hG4bK75ace634
16:17 [562203] From: "0140" <sip:mirs_vs_vld_plsk_2@192.168.1.45>;tag=as486047ec
16:17 [562203] To: <sip:218@192.168.255.103>;tag=bb2963b0-741023
16:17 [562203] Call-ID:65580ace4f6618c75077c9d9282531a8@192.168.1.45
16:17 [562203] CSeq:102 INVITE
16:17 [562203] Contact:<sip:218@192.168.255.103:5060>
16:17 [562203] User-Agent:dlink 12-38-28928749-0.9.5.1.1140-IAD20
16:17 [562203] Content-Type:application/sdp
16:17 [562203] Content-Length:211
16:17 [562203] v=0
16:17 [562203] o=218 1848227130 1848227130 IN IP4 192.168.255.103
16:17 [562203] s=Session SDP
16:17 [562203] c=IN IP4 192.168.255.103
16:17 [562203] t=0 0
16:17 [562203] m=audio 9000 RTP/AVP 8 101
16:17 [562203] a=rtpmap:8 PCMA/8000/1
16:17 [562203] a=rtpmap:101 telephone-event/8000/1
16:17 [562203] a=fmtp:101 0-15
16:17 [562203] -----
16:17 [562205] <---- 192.168.1.45:5060
16:17 [562205] ACK sip:218@192.168.255.103:5060 SIP/2.0
16:17 [562205] Via: SIP/2.0/UDP 192.168.1.45:5060;branch=z9hG4bK7fe53dfa;rport
16:17 [562205] From: "0140" <sip:mirs_vs_vld_plsk_2@192.168.1.45>;tag=as486047ec
16:17 [562205] To: <sip:218@192.168.255.103>;tag=bb2963b0-741023
16:17 [562205] Contact: <sip:mirs_vs_vld_plsk_2@192.168.1.45>
16:17 [562205] Call-ID: 65580ace4f6618c75077c9d9282531a8@192.168.1.45
16:17 [562205] CSeq: 102 ACK
16:17 [562205] User-Agent: Asterisk PBX
16:17 [562205] Max-Forwards: 70
16:17 [562205] Content-Length: 0
16:17 [562205] -----
16:17 [562206] 0: 218=CONNECTED
16:17 [562206] 0: RtpApiTalk[1,1], Peer=192.168.1.45:17362, PT=8, FC=2, NewOOB=1
16:17 [562206] 0: ==14:Talking
19:03 [563867] DSP_ch0_check=0
19:03 [563867] 0: Fxs Hangup
19:03 [563867] 0: Bye at Talking
19:03 [563868] 0: ==3:Idle
19:03 [563868] 0: SetFax(0)=0
19:03 [563868] ----> 192.168.1.45:5060
19:03 [563868] BYE sip:mirs_vs_vld_plsk_2@192.168.1.45 SIP/2.0
19:03 [563868] Via:SIP/2.0/UDP 192.168.255.103:5060;branch=z9hG4bK7c49132943a904e7
19:03 [563868] From: <sip:218@192.168.255.103>;tag=bb2963b0-741023
19:03 [563868] To: "0140" <sip:mirs_vs_vld_plsk_2@192.168.1.45>;tag=as486047ec
19:03 [563868] Call-ID:65580ace4f6618c75077c9d9282531a8@192.168.1.45
19:03 [563868] CSeq:211 BYE
19:04 [563868] Contact:<sip:218@192.168.255.103:5060>
19:04 [563868] Max-Forwards:70
19:04 [563868] User-Agent:dlink 12-38-28928749-0.9.5.1.1140-IAD20
19:04 [563868] Content-Length:0
19:04 [563868] -----
19:04 [563869] <---- 192.168.1.45:5060
19:04 [563869] SIP/2.0 200 OK
19:04 [563869] Via: SIP/2.0/UDP 192.168.255.103:5060;branch=z9hG4bK7c49132943a904e7;received=192.168.255.103
19:04 [563869] From: <sip:218@192.168.255.103>;tag=bb2963b0-741023
19:04 [563869] To: "0140" <sip:mirs_vs_vld_plsk_2@192.168.1.45>;tag=as486047ec
19:04 [563869] Call-ID: 65580ace4f6618c75077c9d9282531a8@192.168.1.45
19:04 [563869] CSeq: 211 BYE
19:04 [563869] User-Agent: Asterisk PBX
19:04 [563869] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
19:04 [563869] Supported: replaces
19:04 [563869] Content-Length: 0
19:04 [563869] -----
19:04 [563870] 0: 218=DISCONNECT
19:04 [563870] 0: Release Old Dlg #2

это входящий разговор. кстати, интересно то, что иногда совершить исходящий вызов все таки получается. Сегодня с утра внезапно само заработало, хотя никто не трогал ничего.
Вот лог успешного исходящего вызова.

Код:

34:42 [573250] 0: 218=REGISTER
34:42 [573251] ----> 192.168.1.45:5060
34:42 [573251] REGISTER sip:192.168.1.45:5060 SIP/2.0
34:42 [573251] Via:SIP/2.0/UDP 192.168.255.103:5060;branch=z9hG4bKb08efca9610a32b5
34:42 [573251] From: "218" <sip:218@192.168.1.45>;tag=5282c164-742137
34:42 [573251] To: "218" <sip:218@192.168.1.45>
34:42 [573251] Call-ID:D1B9-2178-467421377FAC3D134F17-106@SipHost
34:42 [573251] CSeq:216 REGISTER
34:42 [573251] Contact:<sip:218@192.168.255.103:5060>
34:42 [573251] Expires:600
34:42 [573251] Max-Forwards:70
34:42 [573251] User-Agent:dlink 12-38-28928749-0.9.5.1.1140-IAD20
34:42 [573251] Content-Length:0
34:42 [573251] -----
34:42 [573253] <---- 192.168.1.45:5060
34:42 [573253] SIP/2.0 100 Trying
34:42 [573253] Via: SIP/2.0/UDP 192.168.255.103:5060;branch=z9hG4bKb08efca9610a32b5;received=192.168.255.103
34:42 [573253] From: "218" <sip:218@192.168.1.45>;tag=5282c164-742137
34:42 [573253] To: "218" <sip:218@192.168.1.45>
34:42 [573253] Call-ID: D1B9-2178-467421377FAC3D134F17-106@SipHost
34:42 [573253] CSeq: 216 REGISTER
34:42 [573253] User-Agent: Asterisk PBX
34:42 [573253] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
34:42 [573253] Supported: replaces
34:42 [573253] Content-Length: 0
34:42 [573253] -----
34:42 [573253] <---- 192.168.1.45:5060
34:42 [573253] SIP/2.0 401 Unauthorized
34:42 [573253] Via: SIP/2.0/UDP 192.168.255.103:5060;branch=z9hG4bKb08efca9610a32b5;received=192.168.255.103
34:42 [573253] From: "218" <sip:218@192.168.1.45>;tag=5282c164-742137
34:42 [573253] To: "218" <sip:218@192.168.1.45>;tag=as20011fe1
34:42 [573253] Call-ID: D1B9-2178-467421377FAC3D134F17-106@SipHost
34:42 [573253] CSeq: 216 REGISTER
34:42 [573253] User-Agent: Asterisk PBX
34:42 [573253] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
34:42 [573253] Supported: replaces
34:42 [573253] WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3d996352"
34:42 [573253] Content-Length: 0
34:42 [573253] -----
34:42 [573254] 0: 218=REGISTER_AUTHN
34:42 [573254] ----> 192.168.1.45:5060
34:42 [573254] REGISTER sip:192.168.1.45:5060 SIP/2.0
34:42 [573254] Via:SIP/2.0/UDP 192.168.255.103:5060;branch=z9hG4bK5234db1a6ef41a7c
34:42 [573254] From: "218" <sip:218@192.168.1.45>;tag=5282c164-742137
34:42 [573254] To: "218" <sip:218@192.168.1.45>
34:42 [573254] Call-ID:D1B9-2178-467421377FAC3D134F17-106@SipHost
34:42 [573254] CSeq:217 REGISTER
34:43 [573254] Contact:<sip:218@192.168.255.103:5060>
34:43 [573254] Expires:600
34:43 [573254] Max-Forwards:70
34:43 [573254] Authorization:Digest username="218",realm="asterisk",nonce="3d996352",uri="sip:192.168.1.45:5060",response="dfb48c4561587f6e81654d3e8b8b7cf2"
34:43 ,algorithm=MD5
34:43 [573255] User-Agent:dlink 12-38-28928749-0.9.5.1.1140-IAD20
34:43 [573255] Content-Length:0
34:43 [573255] -----
34:43 [573256] <---- 192.168.1.45:5060
34:43 [573256] SIP/2.0 100 Trying
34:43 [573256] Via: SIP/2.0/UDP 192.168.255.103:5060;branch=z9hG4bK5234db1a6ef41a7c;received=192.168.255.103
34:43 [573256] From: "218" <sip:218@192.168.1.45>;tag=5282c164-742137
34:43 [573256] To: "218" <sip:218@192.168.1.45>
34:43 [573256] Call-ID: D1B9-2178-467421377FAC3D134F17-106@SipHost
34:43 [573256] CSeq: 217 REGISTER
34:43 [573256] User-Agent: Asterisk PBX
34:43 [573256] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
34:43 [573256] Supported: replaces
34:43 [573256] Content-Length: 0
34:43 [573256] -----
34:43 [573257] <---- 192.168.1.45:5060
34:43 [573257] SIP/2.0 200 OK
34:43 [573257] Via: SIP/2.0/UDP 192.168.255.103:5060;branch=z9hG4bK5234db1a6ef41a7c;received=192.168.255.103
34:43 [573257] From: "218" <sip:218@192.168.1.45>;tag=5282c164-742137
34:43 [573257] To: "218" <sip:218@192.168.1.45>;tag=as20011fe1
34:43 [573257] Call-ID: D1B9-2178-467421377FAC3D134F17-106@SipHost
34:43 [573257] CSeq: 217 REGISTER
34:43 [573257] User-Agent: Asterisk PBX
34:43 [573257] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
34:43 [573257] Supported: replaces
34:43 [573257] Expires: 600
34:43 [573257] Contact: <sip:218@192.168.255.103:5060>;expires=600
34:43 [573257] Date: Thu, 21 Apr 2011 22:34:39 GMT
34:43 [573257] Content-Length: 0
34:43 [573257] -----
34:43 [573257] 0: 218=REGISTERED
34:54 [573378] 0: Fxs Pickup
34:54 [573378] 0: ==4:GetDtmf
34:58 [573413] 0: DTMF Timeout #1
34:58 [573413] 0: End Input Route [1]
34:58 [573413] 0: Line DialNum_New [343]
34:58 [573413] 0: Call Proxy with [343]
34:58 [573413] 0: ==7:Inviting
34:58 [573413] 0: VoIP CallOut, Invite <sip:343@192.168.1.45:5060>
34:58 [573414] 0: 218=DIALING
34:58 [573415] ----> 192.168.1.45:5060
34:58 [573415] INVITE sip:343@192.168.1.45:5060 SIP/2.0
34:58 [573415] Allow:INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,REFER,SUBSCRIBE,NOTIFY,UPDATE
34:58 [573415] Via:SIP/2.0/UDP 192.168.255.103:5060;branch=z9hG4bK2c5a97eb136791b6
34:58 [573415] From: "218" <sip:218@192.168.1.45>;tag=f7bae384-742153
34:58 [573415] To: <sip:343@192.168.1.45:5060>
34:58 [573415] Call-ID:D1B9-2178-4674215372A555F1E9E9-107@SipHost
34:58 [573415] CSeq:218 INVITE
34:58 [573415] Contact:<sip:218@192.168.255.103:5060>
34:58 [573415] Expires:90
34:58 [573415] Max-Forwards:70
34:58 [573415] Supported:replaces
34:58 [573415] User-Agent:dlink 12-38-28928749-0.9.5.1.1140-IAD20
34:58 [573415] Content-Type:application/sdp
34:58 [573415] Content-Length:237
34:58 [573415] v=0
34:58 [573415] o=218 1849348290 1849348290 IN IP4 192.168.255.103
34:58 [573415] s=Session SDP
34:58 [573415] c=IN IP4 192.168.255.103
34:58 [573415] t=0 0
34:58 [573415] m=audio 9000 RTP/AVP 8 0 101
34:58 [573415] a=rtpmap:8 PCMA/8000/1
34:58 [573415] a=rtpmap:0 PCMU/8000/1
34:58 [573415] a=rtpmap:101 telephone-event/8000/1
34:58 [573415] a=fmtp:101 0-15
34:58 [573415] -----
34:58 [573417] <---- 192.168.1.45:5060
34:58 [573417] SIP/2.0 407 Proxy Authentication Required
34:58 [573417] Via: SIP/2.0/UDP 192.168.255.103:5060;branch=z9hG4bK2c5a97eb136791b6;received=192.168.255.103
34:58 [573417] From: "218" <sip:218@192.168.1.45>;tag=f7bae384-742153
34:58 [573417] To: <sip:343@192.168.1.45:5060>;tag=as7f9489f5
34:58 [573417] Call-ID: D1B9-2178-4674215372A555F1E9E9-107@SipHost
34:58 [573417] CSeq: 218 INVITE
34:59 [573417] User-Agent: Asterisk PBX
34:59 [573417] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
34:59 [573418] Supported: replaces
34:59 [573418] Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="01518ffd"
34:59 [573418] Content-Length: 0
34:59 [573418] -----
34:59 [573419] ----> 192.168.1.45:5060
34:59 [573419] ACK sip:343@192.168.1.45:5060 SIP/2.0
34:59 [573419] Via:SIP/2.0/UDP 192.168.255.103:5060;branch=z9hG4bK2c5a97eb136791b6
34:59 [573419] From: "218" <sip:218@192.168.1.45>;tag=f7bae384-742153
34:59 [573419] To: <sip:343@192.168.1.45:5060>;tag=as7f9489f5
34:59 [573419] Call-ID:D1B9-2178-4674215372A555F1E9E9-107@SipHost
34:59 [573419] CSeq:218 ACK
34:59 [573419] Max-Forwards:70
34:59 [573419] Content-Length:0
34:59 [573419] -----
34:59 [573419] ----> 192.168.1.45:5060
34:59 [573419] INVITE sip:343@192.168.1.45:5060 SIP/2.0
34:59 [573419] Allow:INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,REFER,SUBSCRIBE,NOTIFY,UPDATE
34:59 [573419] Via:SIP/2.0/UDP 192.168.255.103:5060;branch=z9hG4bK38c6ea08b12fb10b
34:59 [573419] From: "218" <sip:218@192.168.1.45>;tag=f7bae384-742153
34:59 [573419] To: <sip:343@192.168.1.45:5060>
34:59 [573419] Call-ID:D1B9-2178-4674215372A555F1E9E9-107@SipHost
34:59 [573419] CSeq:219 INVITE
34:59 [573419] Contact:<sip:218@192.168.255.103:5060>
34:59 [573419] Expires:90
34:59 [573419] Max-Forwards:70
34:59 [573419] Proxy-Authorization:Digest username="218",realm="asterisk",nonce="01518ffd",uri="sip:343@192.168.1.45:5060",response="88ce14d72d32d52ff7810d2
34:59 e44adc7a4",algorithm=MD5
34:59 [573419] Supported:replaces
34:59 [573419] User-Agent:dlink 12-38-28928749-0.9.5.1.1140-IAD20
34:59 [573419] Content-Type:application/sdp
34:59 [573419] Content-Length:237
34:59 [573419] v=0
34:59 [573419] o=218 1849348290 1849348290 IN IP4 192.168.255.103
34:59 [573419] s=Session SDP
34:59 [573419] c=IN IP4 192.168.255.103
34:59 [573419] t=0 0
34:59 [573419] m=audio 9000 RTP/AVP 8 0 101
34:59 [573419] a=rtpmap:8 PCMA/8000/1
34:59 [573419] a=rtpmap:0 PCMU/8000/1
34:59 [573419] a=rtpmap:101 telephone-event/8000/1
34:59 [573419] a=fmtp:101 0-15
34:59 [573419] -----
34:59 [573421] <---- 192.168.1.45:5060
34:59 [573421] SIP/2.0 100 Trying
34:59 [573421] Via: SIP/2.0/UDP 192.168.255.103:5060;branch=z9hG4bK38c6ea08b12fb10b;received=192.168.255.103
34:59 [573421] From: "218" <sip:218@192.168.1.45>;tag=f7bae384-742153
34:59 [573421] To: <sip:343@192.168.1.45:5060>
34:59 [573421] Call-ID: D1B9-2178-4674215372A555F1E9E9-107@SipHost
34:59 [573421] CSeq: 219 INVITE
34:59 [573421] User-Agent: Asterisk PBX
34:59 [573422] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
34:59 [573422] Supported: replaces
34:59 [573422] Contact: <sip:343@192.168.1.45>
34:59 [573422] Content-Length: 0
34:59 [573422] -----
34:59 [573422] 0: 218=PROCEEDING
34:59 [573422] <---- 192.168.1.45:5060
34:59 [573422] SIP/2.0 200 OK
34:59 [573422] Via: SIP/2.0/UDP 192.168.255.103:5060;branch=z9hG4bK38c6ea08b12fb10b;received=192.168.255.103
34:59 [573422] From: "218" <sip:218@192.168.1.45>;tag=f7bae384-742153
34:59 [573422] To: <sip:343@192.168.1.45:5060>;tag=as4cf6d1f5
34:59 [573422] Call-ID: D1B9-2178-4674215372A555F1E9E9-107@SipHost
35:00 [573422] CSeq: 219 INVITE
35:00 [573422] User-Agent: Asterisk PBX
35:00 [573422] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
35:00 [573422] Supported: replaces
35:00 [573422] Contact: <sip:343@192.168.1.45>
35:00 [573422] Content-Type: application/sdp
35:00 [573422] Content-Length: 264
35:00 [573422] v=0
35:00 [573422] o=root 22539 22539 IN IP4 192.168.1.45
35:00 [573422] s=session
35:00 [573422] c=IN IP4 192.168.1.45
35:00 [573422] t=0 0
35:00 [573422] m=audio 17540 RTP/AVP 8 0 101
35:00 [573422] a=rtpmap:8 PCMA/8000
35:00 [573422] a=rtpmap:0 PCMU/8000
35:00 [573423] a=rtpmap:101 telephone-event/8000
35:00 [573423] a=fmtp:101 0-16
35:00 [573423] a=silenceSupp:off - - - -
35:00 [573423] a=ptime:20
35:00 [573423] a=sendrecv
35:00 [573423] -----
35:00 [573424] 0: 218=CONNECTED
35:00 [573424] 0: Peer=192.168.1.45:17540, PT=8, RecvOnly=0
35:00 [573424] 0: RtpApiTalk[1,1], Peer=192.168.1.45:17540, PT=8, FC=2, NewOOB=1
35:00 [573424] 0: ==14:Talking
35:00 [573424] ----> 192.168.1.45:5060
35:00 [573424] ACK sip:343@192.168.1.45 SIP/2.0
35:00 [573424] Via:SIP/2.0/UDP 192.168.255.103:5060;branch=z9hG4bKaf55f7dc10de100f
35:00 [573424] From: "218" <sip:218@192.168.1.45>;tag=f7bae384-742153
35:00 [573424] To: <sip:343@192.168.1.45:5060>;tag=as4cf6d1f5
35:00 [573424] Call-ID:D1B9-2178-4674215372A555F1E9E9-107@SipHost
35:00 [573424] CSeq:219 ACK
35:00 [573424] Max-Forwards:70
35:00 [573424] Proxy-Authorization:Digest username="218",realm="asterisk",nonce="01518ffd",uri="sip:343@192.168.1.45:5060",response="88ce14d72d32d52ff7810d2
35:00 e44adc7a4",algorithm=MD5
35:00 [573425] User-Agent:dlink 12-38-28928749-0.9.5.1.1140-IAD20
35:00 [573425] Content-Length:0
35:00 [573425] -----
35:36 [573800] <---- 192.168.1.45:5060
35:36 [573800] BYE sip:218@192.168.255.103:5060 SIP/2.0
35:37 [573800] Via: SIP/2.0/UDP 192.168.1.45:5060;branch=z9hG4bK1a4df66b;rport
35:37 [573800] From: <sip:343@192.168.1.45:5060>;tag=as4cf6d1f5
35:37 [573800] To: "218" <sip:218@192.168.1.45>;tag=f7bae384-742153
35:37 [573800] Call-ID: D1B9-2178-4674215372A555F1E9E9-107@SipHost
35:37 [573800] CSeq: 102 BYE
35:37 [573800] User-Agent: Asterisk PBX
35:37 [573800] Max-Forwards: 70
35:37 [573800] X-Asterisk-HangupCause: Normal Clearing
35:37 [573800] X-Asterisk-HangupCauseCode: 16
35:37 [573800] Content-Length: 0
35:37 [573800] -----
35:37 [573801] 0: 218=DISCONNECT
35:37 [573801] 0: Release Active Dlg
35:37 [573801] 0: ==15:PlayBusyTone
35:37 [573801] ----> 192.168.1.45:5060
35:37 [573801] SIP/2.0 200 OK
35:37 [573801] Via:SIP/2.0/UDP 192.168.1.45:5060;rport;branch=z9hG4bK1a4df66b
35:37 [573801] From: <sip:343@192.168.1.45:5060>;tag=as4cf6d1f5
35:37 [573801] To: "218" <sip:218@192.168.1.45>;tag=f7bae384-742153
35:37 [573801] Call-ID:D1B9-2178-4674215372A555F1E9E9-107@SipHost
35:37 [573801] CSeq:102 BYE
35:37 [573801] User-Agent:dlink 12-38-28928749-0.9.5.1.1140-IAD20
35:37 [573801] Content-Length:0
35:37 [573801] -----
35:37 [573803] DSP_ch0_check=0
35:37 [573803] 0: Fxs Hangup
35:37 [573803] 0: ==3:Idle
35:37 [573803] 0: SetFax(0)=0
35:42

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Maksim237

Заголовок сообщения: Re: DVG-7062S пропадает pptp связь при SIP вызове

Добавлено: Пт апр 22, 2011 02:01

Зарегистрирован: Пт дек 10, 2010 03:42
Сообщений: 47
Откуда: Хабаровск

внезапно сегодня с утра устойчиво заработало на двух точках, третья пока не проснулась, но думаю, что там тоже все ок.
Вообще, что это могло быть?
Есть еще одна деталь: на всех трех шлюзах дигит мап был настроен таким образом, что внутренние номера шли через FXO.
То есть ситуация:
Настроено все корректно, городские - на город, внутренние на войп. Все хорошо работает. ВНЕЗАПНО, все ломается. Пользователи обращаются с жалобой на невозможность дозвона на внутренние номера.
Выясняется, что внутренние тоже идут на FXO. Я поправляю дигит мап, но дозвониться все равно не получается. Вылезает проблема, с которой я изначально и обратился. Что это все могло быть? Шлюз сам мог подтянуть какие-то обновления с сайта D-Link?

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Alexey Motkov

Заголовок сообщения: Re: DVG-7062S пропадает pptp связь при SIP вызове

Добавлено: Пт апр 22, 2011 09:09

Сотрудник D-LINK

Зарегистрирован: Пн янв 11, 2010 09:40
Сообщений: 4400

Шлюз обновления подтянуть не мог, но в логе шлюз не сигнализировал о том, что именно он разрывает соединение.

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