[AsteriskBrasil] DVG 6004S St_VoipAnswering Timeout
Ivan Maldonado Orosco
ivanorosco em hotmail.com
Quarta Outubro 31 14:41:53 BRST 2012
É meio extenso, segue:
[Oct 30 07:41:20] VERBOSE[18040] chan_sip.c: Audio is at 189.2.20.134 port 16664
[Oct 30 07:41:20] VERBOSE[18040] chan_sip.c: Adding codec 0x4 (ulaw) to SDP
[Oct 30 07:41:20] VERBOSE[18040] chan_sip.c: Adding codec 0x8 (alaw) to SDP
[Oct 30 07:41:20] VERBOSE[18040] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[Oct 30 07:41:20] VERBOSE[18040] chan_sip.c: Reliably Transmitting (no NAT) to 189.47.46.151:5060:
INVITE sip:97095313 em 189.47.46.151:5060 SIP/2.0
Via: SIP/2.0/UDP 189.2.20.134:5060;branch=z9hG4bK4c8b3981;rport
Max-Forwards: 70
From: "100" <sip:100 em 189.2.20.134>;tag=as57399d63
To: <sip:97095313 em 189.47.46.151:5060>
Contact: <sip:100 em 189.2.20.134>
Call-ID: 3abbce8a451fb58b6726e36623e98a89 em 189.2.20.134
CSeq: 102 INVITE
User-Agent: FPBX-2.8.1(1.6.2.17)
Date: Tue, 30 Oct 2012 09:41:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 261
v=0
o=root 1238957108 1238957108 IN IP4 189.2.20.134
s=Asterisk PBX 1.6.2.17
c=IN IP4 189.2.20.134
t=0 0
m=audio 16664 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
---
[Oct 30 07:41:20] VERBOSE[8895] chan_sip.c:
<--- SIP read from UDP:189.47.46.151:5060 --->
SIP/2.0 100 Trying
Via:SIP/2.0/UDP 189.2.20.134:5060;rport;branch=z9hG4bK4c8b3981
From: "100" <sip:100 em 189.2.20.134>;tag=as57399d63
To: <sip:97095313 em 189.47.46.151:5060>
Call-ID:3abbce8a451fb58b6726e36623e98a89 em 189.2.20.134
CSeq:102 INVITE
Content-Type:application/sdp
Content-Length:0
<------------->
[Oct 30 07:41:20] VERBOSE[8895] chan_sip.c: --- (8 headers 0 lines) ---
[Oct 30 07:41:23] VERBOSE[8895] chan_sip.c:
<--- SIP read from UDP:189.47.46.151:5060 --->
SIP/2.0 183 Session in progress
Via:SIP/2.0/UDP 189.2.20.134:5060;rport;branch=z9hG4bK4c8b3981
From: "100" <sip:100 em 189.2.20.134>;tag=as57399d63
To: <sip:97095313 em 189.47.46.151:5060>;tag=8ed4511d-742398
Call-ID:3abbce8a451fb58b6726e36623e98a89 em 189.2.20.134
CSeq:102 INVITE
Contact:<sip:6720 em 189.47.46.151:5060>
User-Agent:dlink 12-37-61926642-0.9.5.1.735
Content-Type:application/sdp
Content-Length:209
v=0
o=6720 1849593660 1849593660 IN IP4 189.47.46.151
s=Session SDP
c=IN IP4 189.47.46.151
t=0 0
m=audio 10000 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-16
<------------->
[Oct 30 07:41:23] VERBOSE[8895] chan_sip.c: --- (10 headers 9 lines) ---
[Oct 30 07:41:23] VERBOSE[8895] chan_sip.c: Found RTP audio format 0
[Oct 30 07:41:23] VERBOSE[8895] chan_sip.c: Found RTP audio format 101
[Oct 30 07:41:23] VERBOSE[8895] chan_sip.c: Found audio description format PCMU for ID 0
[Oct 30 07:41:23] VERBOSE[8895] chan_sip.c: Found audio description format telephone-event for ID 101
[Oct 30 07:41:23] VERBOSE[8895] chan_sip.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
[Oct 30 07:41:23] VERBOSE[8895] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[Oct 30 07:41:23] VERBOSE[8895] chan_sip.c: Peer audio RTP is at port 189.47.46.151:10000
[Oct 30 07:41:32] VERBOSE[8895] chan_sip.c:
<--- SIP read from UDP:189.47.46.151:5060 --->
SIP/2.0 200 OK
Allow:INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,REFER,SUBSCRIBE,NOTIFY,UPDATE
Via:SIP/2.0/UDP 189.2.20.134:5060;rport;branch=z9hG4bK4c8b3981
From: "100" <sip:100 em 189.2.20.134>;tag=as57399d63
To: <sip:97095313 em 189.47.46.151:5060>;tag=8ed4511d-742398
Call-ID:3abbce8a451fb58b6726e36623e98a89 em 189.2.20.134
CSeq:102 INVITE
Contact:<sip:6720 em 189.47.46.151:5060>
User-Agent:dlink 12-37-61926642-0.9.5.1.735
Content-Type:application/sdp
Content-Length:209
v=0
o=6720 1849602720 1849602720 IN IP4 189.47.46.151
s=Session SDP
c=IN IP4 189.47.46.151
t=0 0
m=audio 10000 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-16
<------------->
[Oct 30 07:41:32] VERBOSE[8895] chan_sip.c: --- (11 headers 9 lines) ---
[Oct 30 07:41:32] VERBOSE[8895] chan_sip.c: Found RTP audio format 0
[Oct 30 07:41:32] VERBOSE[8895] chan_sip.c: Found RTP audio format 101
[Oct 30 07:41:32] VERBOSE[8895] chan_sip.c: Found audio description format PCMU for ID 0
[Oct 30 07:41:32] VERBOSE[8895] chan_sip.c: Found audio description format telephone-event for ID 101
[Oct 30 07:41:32] VERBOSE[8895] chan_sip.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
[Oct 30 07:41:32] VERBOSE[8895] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[Oct 30 07:41:32] VERBOSE[8895] chan_sip.c: Peer audio RTP is at port 189.47.46.151:10000
[Oct 30 07:41:32] VERBOSE[8895] chan_sip.c: list_route: hop: <sip:6720 em 189.47.46.151:5060>
[Oct 30 07:41:32] VERBOSE[8895] chan_sip.c: set_destination: Parsing <sip:6720 em 189.47.46.151:5060> for address/port to send to
[Oct 30 07:41:32] VERBOSE[8895] chan_sip.c: set_destination: set destination to 189.47.46.151, port 5060
[Oct 30 07:41:32] VERBOSE[8895] chan_sip.c: Transmitting (no NAT) to 189.47.46.151:5060:
ACK sip:6720 em 189.47.46.151:5060 SIP/2.0
Via: SIP/2.0/UDP 189.2.20.134:5060;branch=z9hG4bK1c6b8c10;rport
Max-Forwards: 70
From: "100" <sip:100 em 189.2.20.134>;tag=as57399d63
To: <sip:97095313 em 189.47.46.151:5060>;tag=8ed4511d-742398
Contact: <sip:100 em 189.2.20.134>
Call-ID: 3abbce8a451fb58b6726e36623e98a89 em 189.2.20.134
CSeq: 102 ACK
User-Agent: FPBX-2.8.1(1.6.2.17)
Content-Length: 0
---
[Oct 30 07:41:32] VERBOSE[8895] chan_sip.c:
<--- SIP read from UDP:189.47.46.151:5060 --->
SIP/2.0 200 OK
Allow:INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,REFER,SUBSCRIBE,NOTIFY,UPDATE
Via:SIP/2.0/UDP 189.2.20.134:5060;rport;branch=z9hG4bK4c8b3981
From: "100" <sip:100 em 189.2.20.134>;tag=as57399d63
To: <sip:97095313 em 189.47.46.151:5060>;tag=8ed4511d-742398
Call-ID:3abbce8a451fb58b6726e36623e98a89 em 189.2.20.134
CSeq:102 INVITE
Contact:<sip:6720 em 189.47.46.151:5060>
User-Agent:dlink 12-37-61926642-0.9.5.1.735
Content-Type:application/sdp
Content-Length:209
v=0
o=6720 1849602720 1849602720 IN IP4 189.47.46.151
s=Session SDP
c=IN IP4 189.47.46.151
t=0 0
m=audio 10000 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-16
<------------->
[Oct 30 07:41:32] VERBOSE[8895] chan_sip.c: --- (11 headers 9 lines) ---
[Oct 30 07:41:34] VERBOSE[8895] chan_sip.c:
<--- SIP read from UDP:189.47.46.151:5060 --->
SIP/2.0 200 OK
Allow:INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,REFER,SUBSCRIBE,NOTIFY,UPDATE
Via:SIP/2.0/UDP 189.2.20.134:5060;rport;branch=z9hG4bK4c8b3981
From: "100" <sip:100 em 189.2.20.134>;tag=as57399d63
To: <sip:97095313 em 189.47.46.151:5060>;tag=8ed4511d-742398
Call-ID:3abbce8a451fb58b6726e36623e98a89 em 189.2.20.134
CSeq:102 INVITE
Contact:<sip:6720 em 189.47.46.151:5060>
User-Agent:dlink 12-37-61926642-0.9.5.1.735
Content-Type:application/sdp
Content-Length:209
v=0
o=6720 1849602720 1849602720 IN IP4 189.47.46.151
s=Session SDP
c=IN IP4 189.47.46.151
t=0 0
m=audio 10000 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-16
<------------->
[Oct 30 07:41:34] VERBOSE[8895] chan_sip.c: --- (11 headers 9 lines) ---
[Oct 30 07:41:36] VERBOSE[8895] chan_sip.c:
<--- SIP read from UDP:189.47.46.151:5060 --->
SIP/2.0 200 OK
Allow:INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,REFER,SUBSCRIBE,NOTIFY,UPDATE
Via:SIP/2.0/UDP 189.2.20.134:5060;rport;branch=z9hG4bK4c8b3981
From: "100" <sip:100 em 189.2.20.134>;tag=as57399d63
To: <sip:97095313 em 189.47.46.151:5060>;tag=8ed4511d-742398
Call-ID:3abbce8a451fb58b6726e36623e98a89 em 189.2.20.134
CSeq:102 INVITE
Contact:<sip:6720 em 189.47.46.151:5060>
User-Agent:dlink 12-37-61926642-0.9.5.1.735
Content-Type:application/sdp
Content-Length:209
v=0
o=6720 1849602720 1849602720 IN IP4 189.47.46.151
s=Session SDP
c=IN IP4 189.47.46.151
t=0 0
m=audio 10000 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-16
<------------->
[Oct 30 07:41:36] VERBOSE[8895] chan_sip.c: --- (11 headers 9 lines) ---
[Oct 30 07:41:40] VERBOSE[8895] chan_sip.c:
<--- SIP read from UDP:189.47.46.151:5060 --->
SIP/2.0 200 OK
Allow:INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,REFER,SUBSCRIBE,NOTIFY,UPDATE
Via:SIP/2.0/UDP 189.2.20.134:5060;rport;branch=z9hG4bK4c8b3981
From: "100" <sip:100 em 189.2.20.134>;tag=as57399d63
To: <sip:97095313 em 189.47.46.151:5060>;tag=8ed4511d-742398
Call-ID:3abbce8a451fb58b6726e36623e98a89 em 189.2.20.134
CSeq:102 INVITE
Contact:<sip:6720 em 189.47.46.151:5060>
User-Agent:dlink 12-37-61926642-0.9.5.1.735
Content-Type:application/sdp
Content-Length:209
v=0
o=6720 1849602720 1849602720 IN IP4 189.47.46.151
s=Session SDP
c=IN IP4 189.47.46.151
t=0 0
m=audio 10000 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-16
<------------->
[Oct 30 07:41:44] VERBOSE[8895] chan_sip.c: --- (10 headers 0 lines) ---
[Oct 30 07:41:44] VERBOSE[8895] chan_sip.c: Really destroying SIP dialog '644015366b7e382f266705505a200615 em 189.2.20.134' Method: OPTIONS
[Oct 30 07:41:48] VERBOSE[8895] chan_sip.c:
<--- SIP read from UDP:189.47.46.151:5060 --->
SIP/2.0 200 OK
Allow:INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,REFER,SUBSCRIBE,NOTIFY,UPDATE
Via:SIP/2.0/UDP 189.2.20.134:5060;rport;branch=z9hG4bK4c8b3981
From: "100" <sip:100 em 189.2.20.134>;tag=as57399d63
To: <sip:97095313 em 189.47.46.151:5060>;tag=8ed4511d-742398
Call-ID:3abbce8a451fb58b6726e36623e98a89 em 189.2.20.134
CSeq:102 INVITE
Contact:<sip:6720 em 189.47.46.151:5060>
User-Agent:dlink 12-37-61926642-0.9.5.1.735
Content-Type:application/sdp
Content-Length:209
v=0
o=6720 1849602720 1849602720 IN IP4 189.47.46.151
s=Session SDP
c=IN IP4 189.47.46.151
t=0 0
m=audio 10000 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-16
<------------->
[Oct 30 07:41:48] VERBOSE[8895] chan_sip.c: --- (11 headers 9 lines) ---
[Oct 30 07:41:53] WARNING[18042] func_db.c: DB_DELETE requires an argument, DB_DELETE(<family>/<key>)
[Oct 30 07:41:53] VERBOSE[8895] chan_sip.c:
<--- SIP read from UDP:189.47.46.151:5060 --->
BYE sip:100 em 189.2.20.134 SIP/2.0
Via:SIP/2.0/UDP 189.47.46.151:5060;branch=z9hG4bK567b6611820f8f0c
From: <sip:97095313 em 189.47.46.151:5060>;tag=8ed4511d-742398
To: "100" <sip:100 em 189.2.20.134>;tag=as57399d63
Call-ID:3abbce8a451fb58b6726e36623e98a89 em 189.2.20.134
CSeq:8 BYE
Contact:<sip:6720 em 189.47.46.151:5060>
Max-Forwards:70
User-Agent:dlink 12-37-61926642-0.9.5.1.735
Content-Length:0
<------------->
[Oct 30 07:41:53] VERBOSE[8895] chan_sip.c: --- (10 headers 0 lines) ---
[Oct 30 07:41:53] VERBOSE[8895] chan_sip.c: Sending to 189.47.46.151 : 5060 (no NAT)
[Oct 30 07:41:53] VERBOSE[8895] chan_sip.c:
<--- Transmitting (no NAT) to 189.47.46.151:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 189.47.46.151:5060;branch=z9hG4bK567b6611820f8f0c;received=189.47.46.151
From: <sip:97095313 em 189.47.46.151:5060>;tag=8ed4511d-742398
To: "100" <sip:100 em 189.2.20.134>;tag=as57399d63
Call-ID: 3abbce8a451fb58b6726e36623e98a89 em 189.2.20.134
CSeq: 8 BYE
Server: FPBX-2.8.1(1.6.2.17)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
<------------>
[Oct 30 07:41:54] VERBOSE[8895] chan_sip.c: Really destroying SIP dialog '3abbce8a451fb58b6726e36623e98a89 em 189.2.20.134' Method: BYE
[Oct 30 07:42:15] VERBOSE[8895] chan_sip.c: Really destroying SIP dialog 'CE40-3FA8-4674241856E1641BE1B1-193 em SipHost' Method: REGISTER
[Oct 30 07:42:16] VERBOSE[8895] chan_sip.c: Really destroying SIP dialog 'CE40-3FA8-4674242047F2EB3729E8-194 em SipHost' Method: REGISTER
[]’s
From: Ivan Paes José
Sent: Tuesday, October 30, 2012 12:20 AM
To: asteriskbrasil em listas.asteriskbrasil.org
Subject: Re: [AsteriskBrasil] DVG 6004S St_VoipAnswering Timeout
Olá!
Consegues o debug do sip do CLI do asterisk?
Atenciosamente,
Ivan Paes José
Acadêmico de Biblioteconomia - UFSC
Técnico em Telecomunicações - IFSC
E-mail/MSN/GTalk: ivan.paes em gmail.com
Oi: +55 48 84291055
Skype: ivanpaesjose
Palhoça - Santa Catarina - Brasil
*** Muito Importante *** NETiqueta
Se repassar esta mensagem, por gentileza:
* Apague todos os endereços que aparecem nele.
* E, por opção de segurança endereçá-lo no Cco ou Bcc.
Desta forma você estará protegendo a mim, seus amigos e a você mesmo.
Eu, juntamente com a campanha contra a propagação de vírus agradecemos sinceramente.
Em 29 de outubro de 2012 17:28, Ivan Maldonado Orosco <ivanorosco em hotmail.com> escreveu:
Boa tarde,
Adquirimos um ata Dlink DVG 6004S e configuramos o mesmo para utilizar nossas linhas analógicas, no entanto, deparamos com um problema aparentemente simples para se resolver, no entanto, não conseguimos encontrar a solução. O que acontece é que o ata recebe ligações e transfere para os ramais SIP corretamente (entrada 100% funcional), mas na execução de discagem, o asterisk comanda a discagem, o ata faz a discagem, o destino atende a ligação e os dois pontos se falam, mas após 20 segundos da discagem a ligação é derrubada pelo ata.
Depurando as ações dos comandos do ata pelo SLmon (programinha da Dlink), vimos que o mesmo efetua todo o processo de discagem e envia por último o comando “==13: VoipAnswering” (com a ligação já atendida) e após 20 segundos ele responde novamente com St_VoipAnswering Timeout e derruba a ligação.
Vejam o log gerado:
16:58:00 [010667] 5: 6720=OFFERING
16:58:00 [010667] 5: Get CallerId=100
16:58:00 [010668] 5: Check Trunk FixLine
16:58:00 [010668] 5: Hunting Trunk Line
16:58:00 [010668] 0: Peer PTime=20 #2
16:58:00 [010668] 0: Peer=189.2.20.138:19292, PT=0, RecvOnly=0
16:58:00 [010668] 0: TrunkPrefix=, Dest=39064886, Dialno=39064886
16:58:00 [010668] 0: FxoHookOff
16:58:00 [010668] 0: SetInputGain(-2)
16:58:00 [010668] 0: ==18:TrunkDialOut
16:58:01 [010679] 0: DialOut(39064886)=0
16:58:02 [010692] 0: RtpApiTalk[1,1],Peer=189.2.20.138:19292,PT=0,FX=2,NewOOB=1
16:58:02 [010692] 0: Substatus=3
16:58:11 [010782] 0: Fxo Still No RingTone, Talk
16:58:11 [010782] 0: Fxo DialOut OK
16:58:11 [010782] 5: 6720=ACCEPT
16:58:11 [010783] 0: ==13:VoipAnswering
16:58:33 [010996] 0: St_VoipAnswering Timeout
16:58:33 [010996] 0: DSP_ch0_check=0
16:58:33 [010996] 0: FxoHookOn
16:58:33 [010996] 0: ==15:PlayBusyTone
16:58:33 [010996] 0: DSP_ch0_check=0
16:58:33 [010996] 0: FxoHookOn
16:58:33 [010996] 0: ==3:Idle
16:58:33 [010996] 0: SetInputGain(4)
16:58:33 [010996] 0: SetFax(1)=0
16:58:33 [010996] 0: 6720=DISCONNECT
Já fiz todas as configurações possíveis no ATA e não houve nenhum resultado diferente que faça ele dar uma outra resposta após o VoipAnswering que não seja Timeout.
As configurações de meu tronco SIP são:
host=dynamic
username=6720
secret=
type=friend
qualify=yes
canreinvite=yes
dtmfmode=rfc2833
alow=all
O ATA está ligado diretamente na internet sem firewal na frente, assim como nosso asterisk (para evitar qualquer problema relacionado a liberação de portas)
Alguém tem alguma idéia ?
[]’s
_______________________________________________
KHOMP Inovação: External Board Series
Módulos de 1/2 rack e 1U para todas as interfaces e soluções Asterisk e FreeSWITCH.
Tenha a External Series Experience na sua aplicação. Visite www.khomp.com
_______________________________________________
DIGIVOICE Fabricante de Placas de Voz e Channel Bank
20 anos de experiência com E1(R2/ISDN), FXS, FXO e GSM
Centro Treinamento - Curso de PABX IP - Asterisk - Site www.digivoice.com.br
________
YEALINK: Telefones IP e VídeoPhones IP com o melhor custo/benefício do mercado.
email: yealink em commlogik.com.br | www.commlogik.com.br | (11) 5503-1011
______________________________________________
Para remover seu email desta lista, basta enviar um email em branco para asteriskbrasil-unsubscribe em listas.asteriskbrasil.org
--------------------------------------------------------------------------------
_______________________________________________
KHOMP Inovação: External Board Series
Módulos de 1/2 rack e 1U para todas as interfaces e soluções Asterisk e FreeSWITCH.
Tenha a External Series Experience na sua aplicação. Visite www.khomp.com
_______________________________________________
DIGIVOICE Fabricante de Placas de Voz e Channel Bank
20 anos de experiência com E1(R2/ISDN), FXS, FXO e GSM
Centro Treinamento - Curso de PABX IP - Asterisk - Site www.digivoice.com.br
________
YEALINK: Telefones IP e VídeoPhones IP com o melhor custo/benefício do mercado.
email: yealink em commlogik.com.br | www.commlogik.com.br | (11) 5503-1011
______________________________________________
Para remover seu email desta lista, basta enviar um email em branco para asteriskbrasil-unsubscribe em listas.asteriskbrasil.org
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