[AsteriskBrasil] Problema com Asterisk 1.8

Sexta Junho 17 21:59:16 BRT 2011

Amigo, vc só esqueceu de remover usuário, senha e IP de autenticação do seu
servidor....

Em 17 de junho de 2011 16:50, Emerson Mochon Borsatti <borsatti.em em gmail.com
> escreveu:

> Boa tarde pessoal,
>
> Estou com um problema com meu asterisk 1.8 rodando no Ubuntu. Estava tudo
> funcionando perfeitamente há varios meses, mas tive de dar um boot na
> máquina, estou com vários problemas.
> Usuários internos à rede, ligam e recebem ligação sem problema. Agora *quem
> esta externo (conecta de um ip público) ouve mas não consegue falar* (a
> outra ponta nãp recebe o áudio).
>
> O que pode ser?
>
> Já testei o nat como comedia, force_rport, yes, no , tb já
> testei directmedia=no ; directrtpsetup=yes e no
>
> Eis algumas configurações:
>
> [3124]
> CID_3124 = XXXX3124
> type=friend
> context=imti-broad
> callerid=Alguem cel <3124>
> secret=3124
> host=dynamic
> nat=yes
> directmedia=no
> directrtpsetup=yes
> mailbox=
> disallow=all
> allow=ulaw
> allow=alaw
> allow=gsm
> allow=g729
> allow=g723.1
> allow=ilbc
> qualify=yes
> amaflags=billing
>
>
> Alguns trechos do sip debug:
>
> <------------->
>
> --- (15 headers 12 lines) ---
>
> Sending to 189.99.147.248:50229 (no NAT)
>
> Using INVITE request as basis request -
> 98880C4AF1C2D5A495758A0536857027ECB5C605
>
> Found peer '3124' for '3124' from 189.99.147.248:50229
>
>
>
> <--- Reliably Transmitting (no NAT) to 189.99.147.248:50229 --->
>
> SIP/2.0 401 Unauthorized
>
> Via: SIP/2.0/UDP 189.99.147.248:50229
> ;branch=z9hG4bKNYHl4TEKjJOEdah1;received=189.99.147.248;rport=50229
>
> From: "Napa" <sip:3124 em voip.capital.ms.gov.br
> >;tag=F653FF04FBC97E9DDA8F463736A1EACD
>
> To: <sip:091012026 em voip.capital.ms.gov.br>;tag=as4b635a6b
>
> Call-ID: 98880C4AF1C2D5A495758A0536857027ECB5C605
>
> CSeq: 1 INVITE
>
> Server: Asterisk PBX 1.8.0
>
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
> PUBLISH
>
> Supported: replaces, timer
>
> WWW-Authenticate: Digest algorithm=MD5, realm="", nonce="3b9b9f7a"
>
> Content-Length: 0
>
>
>
>
>
> <------------>
>
> Scheduling destruction of SIP dialog
> '98880C4AF1C2D5A495758A0536857027ECB5C605' in 86400 ms (Method: INVITE)
>
>
>
> <--- SIP read from UDP:189.99.147.248:50229 --->
>
>
> <--- SIP read from UDP:172.17.0.27:5060 --->
>
> SIP/2.0 180 Ringing
>
> Via: SIP/2.0/UDP 172.17.0.30:5060
> ;branch=z9hG4bK63996783;received=172.17.0.30
>
> From: "Napa cel" <sip:33043124 em 172.17.0.30>;tag=as5f695838
>
> To: <sip:91012026 em 172.17.0.27>;tag=as0c5ebd9a
>
> Call-ID: 4b3c1ec75a15b5055f6135bd5532e081 em 172.17.0.30:5060
>
> CSeq: 102 INVITE
>
> Server: Asterisk PBX 1.6.2.14
>
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
>
> Supported: replaces, timer
>
> Contact: <sip:91012026 em 172.17.0.27>
>
> Content-Length: 0
>
>
>
>
>
> <------------->
>
>
> <------------->
>
> Reliably Transmitting (no NAT) to 189.99.147.248:50229:
>
> OPTIONS sip:3124 em 189.99.147.248:50229;rinstance=C0E3D765 SIP/2.0
>
> Via: SIP/2.0/UDP 172.17.0.30:5060;branch=z9hG4bK63c167cf
>
> Max-Forwards: 30
>
> From: "asterisk" <sip:asterisk em 172.17.0.30>;tag=as36b7d17f
>
> To: <sip:3124 em 189.99.147.248:50229;rinstance=C0E3D765>
>
> Contact: <sip:asterisk em 172.17.0.30:5060>
>
> Call-ID: 11fdeb005a975be12673b175417d892d em 172.17.0.30:5060
>
> CSeq: 102 OPTIONS
>
> User-Agent: Asterisk PBX 1.8.0
>
> Date: Fri, 17 Jun 2011 19:09:19 GMT
>
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
> PUBLISH
>
> Supported: replaces, timer
>
> Content-Length: 0
>
>
>
>
>
> ---
>
> Really destroying SIP dialog 'pSaqJ-7hH0441f2 em capital.ms.gov.br' Method:
> REGISTER
>
> Retransmitting #1 (no NAT) to 189.99.147.248:50229:
>
> OPTIONS sip:3124 em 189.99.147.248:50229;rinstance=C0E3D765 SIP/2.0
>
> Via: SIP/2.0/UDP 172.17.0.30:5060;branch=z9hG4bK63c167cf
>
> Max-Forwards: 30
>
> From: "asterisk" <sip:asterisk em 172.17.0.30>;tag=as36b7d17f
>
> To: <sip:3124 em 189.99.147.248:50229;rinstance=C0E3D765>
>
> Contact: <sip:asterisk em 172.17.0.30:5060>
>
> Call-ID: 11fdeb005a975be12673b175417d892d em 172.17.0.30:5060
>
> CSeq: 102 OPTIONS
>
> User-Agent: Asterisk PBX 1.8.0
>
> Date: Fri, 17 Jun 2011 19:09:19 GMT
>
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
> PUBLISH
>
> Supported: replaces, timer
>
> Content-Length: 0
>
>
>
>
>
> ---
>
> Really destroying SIP dialog '12743f7a63a87dae3acfb44f781f59a5 em 172.17.0.27'
> Method: ACK
>
>
>
> <--- SIP read from UDP:10.4.0.152:5060 --->
>
>
>
>
>
> <------------->
>
>
>
> <--- SIP read from UDP:189.99.147.248:50229 --->
>
> SIP/2.0 200 OK
>
> From: "asterisk" <sip:asterisk em 172.17.0.30>;tag=as36b7d17f
>
> Call-ID: 11fdeb005a975be12673b175417d892d em 172.17.0.30:5060
>
> CSeq: 102 OPTIONS
>
> Via: SIP/2.0/UDP 172.17.0.30:5060;branch=z9hG4bK63c167cf
>
> To: <sip:3124 em 189.99.147.248:50229;rinstance=C0E3D765>
>
> Contact: <sip:3124 em 189.99.147.248:50229>
>
> Allow: OPTIONS, INVITE, ACK, REFER, CANCEL, BYE, NOTIFY
>
> Supported: replaces
>
> Supported: path
>
> Accept: application/sdp
>
> Content-Length: 0
>
>
>
>
>
> <------------->
>
> --- (12 headers 0 lines) ---
>
> Really destroying SIP dialog '
> 11fdeb005a975be12673b175417d892d em 172.17.0.30:5060' Method: OPTIONS
>
>
>
> <--- SIP read from UDP:172.17.0.27:5060 --->
>
> SIP/2.0 200 OK
>
> Via: SIP/2.0/UDP 172.17.0.30:5060
> ;branch=z9hG4bK63996783;received=172.17.0.30
>
> From: "Napa cel" <sip:33043124 em 172.17.0.30>;tag=as5f695838
>
> To: <sip:91012026 em 172.17.0.27>;tag=as0c5ebd9a
>
> Call-ID: 4b3c1ec75a15b5055f6135bd5532e081 em 172.17.0.30:5060
>
> CSeq: 102 INVITE
>
> Server: Asterisk PBX 1.6.2.14
>
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
>
> Supported: replaces, timer
>
> Contact: <sip:91012026 em 172.17.0.27>
>
> Content-Type: application/sdp
>
> Content-Length: 331
>
>
>
> v=0
>
> o=root 810974455 810974455 IN IP4 172.17.0.27
>
> s=Asterisk PBX 1.6.2.14
>
> c=IN IP4 172.17.0.27
>
> t=0 0
>
> m=audio 10922 RTP/AVP 18 0 8 101
>
> a=rtpmap:18 G729/8000
>
> a=fmtp:18 annexb=no
>
> a=rtpmap:0 PCMU/8000
>
> a=rtpmap:8 PCMA/8000
>
> a=rtpmap:101 telephone-event/8000
>
> a=fmtp:101 0-16
>
> a=silenceSupp:off - - - -
>
> a=ptime:20
>
> a=sendrecv
>
>
>
> <------------->
>
> --- (12 headers 15 lines) ---
>
> Found RTP audio format 18
>
> Found RTP audio format 0
>
> Found RTP audio format 8
>
> Found RTP audio format 101
>
> Found audio description format G729 for ID 18
>
> Found audio description format PCMU for ID 0
>
> Found audio description format PCMA for ID 8
>
> Found audio description format telephone-event for ID 101
>
> Capabilities: us - 0x10f (g723|gsm|ulaw|alaw|g729), peer - audio=0x10c
> (ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x10c
> (ulaw|alaw|g729)
>
> Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1
> (telephone-event|), combined - 0x1 (telephone-event|)
>
> Peer audio RTP is at port 172.17.0.27:10922
>
> list_route: hop: <sip:91012026 em 172.17.0.27>
>
> set_destination: Parsing <sip:91012026 em 172.17.0.27> for address/port to
> send to
>
> set_destination: set destination to 172.17.0.27:5060
>
> Transmitting (no NAT) to 172.17.0.27:5060:
>
> ACK sip:91012026 em 172.17.0.27 SIP/2.0
>
> Via: SIP/2.0/UDP 172.17.0.30:5060;branch=z9hG4bK4eb008f1
>
> Max-Forwards: 30
>
> From: "Napa cel" <sip:33043124 em 172.17.0.30>;tag=as5f695838
>
> To: <sip:91012026 em 172.17.0.27>;tag=as0c5ebd9a
>
> Contact: <sip:33043124 em 172.17.0.30:5060>
>
> Call-ID: 4b3c1ec75a15b5055f6135bd5532e081 em 172.17.0.30:5060
>
> CSeq: 102 ACK
>
> User-Agent: Asterisk PBX 1.8.0
>
> Content-Length: 0
>
>
>
>
>
> ---
>
>     -- SIP/voipe1-000001d8 answered SIP/3124-000001d7
>
> Audio is at 5060
>
> Adding codec 0x4 (ulaw) to SDP
>
> Adding codec 0x8 (alaw) to SDP
>
> Adding codec 0x2 (gsm) to SDP
>
> Adding codec 0x400 (ilbc) to SDP
>
> Adding non-codec 0x1 (telephone-event) to SDP
>
>
>
> <--- Reliably Transmitting (no NAT) to 189.99.147.248:50229 --->
>
> SIP/2.0 200 OK
>
> Via: SIP/2.0/UDP 189.99.147.248:50229
> ;branch=z9hG4bKErQ5G8Qstb3cnCaI;received=189.99.147.248;rport=50229
>
> From: "Napa" <sip:3124 em voip.capital.ms.gov.br
> >;tag=F653FF04FBC97E9DDA8F463736A1EACD
>
> To: <sip:091012026 em voip.capital.ms.gov.br>;tag=as63c3ce2e
>
> Call-ID: 98880C4AF1C2D5A495758A0536857027ECB5C605
>
> CSeq: 2 INVITE
>
> Server: Asterisk PBX 1.8.0
>
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
> PUBLISH
>
> Supported: replaces, timer
>
> Contact: <sip:091012026 em 172.17.0.30:5060>
>
> Content-Type: application/sdp
>
> Content-Length: 327
>
>
>
> v=0
>
> o=root 1704978075 1704978075 IN IP4 172.17.0.30
>
> s=Asterisk PBX 1.8.0
>
> c=IN IP4 172.17.0.30
>
> t=0 0
>
> m=audio 12602 RTP/AVP 0 8 3 102 101
>
> a=rtpmap:0 PCMU/8000
>
> a=rtpmap:8 PCMA/8000
>
> a=rtpmap:3 GSM/8000
>
> a=rtpmap:102 iLBC/8000
>
> a=fmtp:102 mode=30
>
> a=rtpmap:101 telephone-event/8000
>
> a=fmtp:101 0-16
>
> a=ptime:20
>
> a=sendrecv
>
>
>
> <------------>
>
>     -- Locally bridging SIP/3124-000001d7 and SIP/voipe1-000001d8
>
>
> --
> Att.,
>
> *Emerson M. Borsatti*
> Mandriva Linux System Administrator Certified
> ITIL V3 Intermediate Certified
> ISO 20000 Certified
> *(67) 9101-2026*
>
>
> _______________________________________________
> KHOMP: qualidade em placas de E1, GSM, FXS e FXO para Asterisk.
> - Hardware com alta disponibilidade de recursos e qualidade KHOMP
> - Suporte técnico local qualificado e gratuito
> Conheça a linha completa de produtos KHOMP em www.khomp.com.br
> _______________________________________________
> DIGIVOICE: Lider no mercado de placas para Asterisk
> Único fabricante com Centro de Treinamento especializado.
> LANÇAMENTO: Channel Bank TDMoE, até 64 canais FXS / FXO.
> www.digivoice.com.br ou (11)3016-5200.
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-- 
Att.,

Leandro Alves
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