[AsteriskBrasil] DVG 6004S St_VoipAnswering Timeout

Domingo Novembro 4 16:30:07 BRST 2012

Não cheguei monitorar. Apenas precisa de uma solução rápida e fiz da 
forma que lhe mostrei e está funcionando até hoje.

Em 01/11/2012 09:16, Ivan Maldonado Orosco escreveu:
> Bom, ao menos é um caminho para eu pesquisar, mas Guilherme, o seu 
> problema era exatamente o mesmo ? você chegou a monitorar o seu dvg 
> 6004/6008 e viu esse bemdito VoipAnswering e St_VoipAnswering Timeout ?
> []'s
> *From:* Guilherme Rezende <mailto:asterisk em guilherme.eti.br>
> *Sent:* Wednesday, October 31, 2012 4:46 PM
> *To:* asteriskbrasil em listas.asteriskbrasil.org 
> <mailto:asteriskbrasil em listas.asteriskbrasil.org>
> *Subject:* Re: [AsteriskBrasil] DVG 6004S St_VoipAnswering Timeout
> Ichi, como uso Asterisk Puro, não vou saber te ajudar.  Mas essa 
> solução que encontrei foi essa  e resolveu meu problema.  Terás que 
> "futucar" no DialPlan do Asterisk.
> Abs..
> Em 31/10/2012 14:41, Ivan Maldonado Orosco escreveu:
>> 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 
>> <mailto: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 
>> <mailto: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 mailto:%27644015366b7e382f266705505a200615 em 189.2.20.134%27 
>> 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 
>> <mailto: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 mailto:%273abbce8a451fb58b6726e36623e98a89 em 189.2.20.134%27 
>> Method: BYE
>> [Oct 30 07:42:15] VERBOSE[8895] chan_sip.c: Really destroying SIP 
>> dialog mailto:%27CE40-3FA8-4674241856E1641BE1B1-193 em SipHost%27 
>> Method: REGISTER
>> [Oct 30 07:42:16] VERBOSE[8895] chan_sip.c: Really destroying SIP 
>> dialog mailto:%27CE40-3FA8-4674242047F2EB3729E8-194 em SipHost%27 
>> Method: REGISTER
>> []'s
>> *From:* Ivan Paes José <mailto:ivan.paes em gmail.com>
>> *Sent:* Tuesday, October 30, 2012 12:20 AM
>> *To:* asteriskbrasil em listas.asteriskbrasil.org 
>> <mailto: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 <mailto: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 <mailto: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
>>     <http://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
>>     <http://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 <http://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 <http://www.digivoice.com.br>
>>     ________
>>     YEALINK: Telefones IP e VídeoPhones IP com o melhor
>>     custo/benefício do mercado.
>>     email: yealink em commlogik.com.br <mailto:yealink em commlogik.com.br>
>>     | www.commlogik.com.br <http://www.commlogik.com.br> | (11)
>>     5503-1011 <tel:%2811%29%205503-1011>
>>     ______________________________________________
>>     Para remover seu email desta lista, basta enviar um email em
>>     branco para asteriskbrasil-unsubscribe em listas.asteriskbrasil.org
>>     <mailto: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
>>
>>
>> _______________________________________________
>> 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. Visitewww.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  - Sitewww.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 paraasteriskbrasil-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
>
>
> _______________________________________________
> 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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