[AsteriskBrasil] DVG 6004S St_VoipAnswering Timeout
Guilherme Rezende
asterisk em guilherme.eti.br
Quarta Outubro 31 16:46:19 BRST 2012
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 '644015366b7e382f266705505a200615 em 189.2.20.134'
> <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 '3abbce8a451fb58b6726e36623e98a89 em 189.2.20.134'
> <mailto:%273abbce8a451fb58b6726e36623e98a89 em 189.2.20.134%27> Method: BYE
> [Oct 30 07:42:15] VERBOSE[8895] chan_sip.c: Really destroying SIP
> dialog 'CE40-3FA8-4674241856E1641BE1B1-193 em SipHost'
> <mailto:%27CE40-3FA8-4674241856E1641BE1B1-193 em SipHost%27> Method: REGISTER
> [Oct 30 07:42:16] VERBOSE[8895] chan_sip.c: Really destroying SIP
> dialog 'CE40-3FA8-4674242047F2EB3729E8-194 em SipHost'
> <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. 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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