[AsteriskBrasil] [Pacotes SIP] Problema com Telefone Voiper da Intelbrás
Diogo Luz
dogoluz em gmail.com
Quarta Março 18 16:12:55 BRT 2009
O seu asterisk está tentando fazer a troca de CODECs e nao esta conseguindo
todos tem que esta Habilitados no seu TElefone IP Voiper.
Configurando Codec exemplo : e os outros
*Codec Preferido* Codec Preferido 1: G.711 lei-u G.711 lei-a G.723
G.729 G.726 - 16 G.726 - 24 G.726 - 32 G.726 - 40 GSM Codec Preferido 2: Não
usado G.711 lei-u G.711 lei-a G.723 G.729 G.726 - 16 G.726 - 24 G.726
- 32 G.726
- 40 GSM
2009/3/18 Junior Polegato - Asterisk <asterisk em juniorpolegato.com.br>
> Olá,
>
> Criei um novo ambiente e consegui fazer refazer a situação do
> problema: quando faço uma ligação e esta não é atendida e coloco no
> gancho, acontece o problema, com essa troca de pacotes:
>
> <-- SIP read from 10.1.1.101:5060:
> Via: SIP/2.0/UDP :5060
> From: "Voiper - 2222" <sip:2222 em asterisk>;tag=3828945f
> To: <sip:3941xxxx em asterisk>;tag=as1c30ec09
> Call-ID: 5d50123233dac07a5e92a42e6d58f883 em 192.168.0.12
> CSeq: 802 ACK
> Content-Length: 0
>
> --- (6 headers 0 lines) ---
> Mar 18 15:25:41 NOTICE[27276]: chan_sip.c:3989 copy_via_headers: No
> header field 'Via' present to copy
> Transmitting (NAT) to 10.1.1.101:5060:
> SIP/2.0 481 Call leg/transaction does not exist
> From: "Voiper - 2222" <sip:2222 em asterisk>;tag=3828945f
> To: <sip:3941xxxx em asterisk>;tag=as1c30ec09
> Call-ID: 5d50123233dac07a5e92a42e6d58f883 em 192.168.0.12
> CSeq: 802 ACK
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Content-Length: 0
>
> Isso prossegue indefinidamente até que se reinicie o aparelho ou
> o Asterisk. Durante isso, nem consigo acesso via Web à configuração do
> aparelho. Reiniciando o Asterisk, imediatamente já aprece a tela de
> login no navegador.
>
> Os pacotes de quando o problema inicia:
>
>
> -- SIP/saida_azzu-081c88a0 is making progress passing it to
> SIP/22-0817f268
> We're at 10.1.1.254 port 11504
> Adding codec 0x100 (g729) to SDP
> Adding codec 0x2 (gsm) to SDP
> Adding non-codec 0x1 (telephone-event) to SDP
> Transmitting (NAT) to 10.1.1.101:5060:
> SIP/2.0 183 Session Progress
> Via: SIP/2.0/UDP
> 10.1.1.101:5060;branch=z9hG4bK36da28d784;received=10.1.1.101;rport=5060
> From: "Voiper - 2222" <sip:2222 em asterisk>;tag=5696a690
> To: <sip:3941xxxx em asterisk>;tag=as5771641a
> Call-ID: 628033914eb9a71c7e8a47a42865a0e3 em 10.1.1.101
> CSeq: 802 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Contact: <sip:3941xxxx em 10.1.1.254 <sip%3A3941xxxx em 10.1.1.254>>
> Content-Type: application/sdp
> Content-Length: 264
>
> v=0
> o=root 8289 8289 IN IP4 10.1.1.254
> s=session
> c=IN IP4 10.1.1.254
> t=0 0
> m=audio 11504 RTP/AVP 18 3 101
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:3 GSM/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
>
> ---
> 12 headers, 0 lines
> Reliably Transmitting (NAT) to 10.1.1.101:5060:
> OPTIONS sip:2222 em 10.1.1.101:5060 SIP/2.0
> Via: SIP/2.0/UDP 10.1.1.254:5060;branch=z9hG4bK172765da;rport
> From: "asterisk" <sip:asterisk em 10.1.1.254 <sip%3Aasterisk em 10.1.1.254>
> >;tag=as027fc344
> To: <sip:2222 em 10.1.1.101:5060>
> Contact: <sip:asterisk em 10.1.1.254 <sip%3Aasterisk em 10.1.1.254>>
> Call-ID: 5a32290f259e06f65e35145944dd7605 em 10.1.1.254
> CSeq: 102 OPTIONS
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Date: Wed, 18 Mar 2009 18:39:10 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Content-Length: 0
>
>
> ---
> testador*CLI>
> <-- SIP read from 10.1.1.101:5060:
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP
> 10.1.1.254:5060;rport=5060;received=10.1.1.254;branch=z9hG4bK172765da
> From: "asterisk" <sip:asterisk em 10.1.1.254 <sip%3Aasterisk em 10.1.1.254>
> >;tag=as027fc344
> To: <sip:2222 em 10.1.1.101:5060>;tag=6ff92d4b
> Call-ID: 5a32290f259e06f65e35145944dd7605 em 10.1.1.254
> Contact: <sip:2222 em 10.1.1.101:5060>
> CSeq: 102 OPTIONS
> Allow: INVITE,CANCEL,ACK,BYE,NOTIFY,REFER,OPTIONS
> Content-Length: 0
>
> --- (9 headers 0 lines) ---
> Destroying call '5a32290f259e06f65e35145944dd7605 em 10.1.1.254'
> testador*CLI>
> <-- SIP read from 10.1.1.101:5060:
>
> --- (0 headers 0 lines) Nat keepalive ---
> testador*CLI>
> <-- SIP read from 10.1.1.101:5060:
>
> --- (0 headers 0 lines) Nat keepalive ---
> testador*CLI>
> <-- SIP read from 10.1.1.101:5060:
> CANCEL sip:3941xxxx em asterisk SIP/2.0
> Via: SIP/2.0/UDP 10.1.1.101:5060;rport;branch=z9hG4bK36da28d784
> From: "Voiper - 2222" <sip:2222 em asterisk>;tag=5696a690
> To: <sip:3941xxxx em asterisk>;tag=as5771641a
> Call-ID: 628033914eb9a71c7e8a47a42865a0e3 em 10.1.1.101
> Contact: <sip:2222 em 10.1.1.101:5060>
> CSeq: 802 CANCEL
> Proxy-Authorization: Digest
>
> username="2222",realm="yyyyyyyyyyyy",nonce="12345678",response="xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx",uri="sip:3941xxxx em asterisk
> ",algorithm=MD5
> Content-Length: 0
>
> --- (9 headers 0 lines) ---
> Sending to 10.1.1.101 : 5060 (NAT)
> Reliably Transmitting (NAT) to 10.1.1.101:5060:
> SIP/2.0 487 Request Terminated
> Via: SIP/2.0/UDP
> 10.1.1.101:5060;branch=z9hG4bK36da28d784;received=10.1.1.101;rport=5060
> From: "Voiper - 2222" <sip:2222 em asterisk>;tag=5696a690
> To: <sip:3941xxxx em asterisk>;tag=as5771641a
> Call-ID: 628033914eb9a71c7e8a47a42865a0e3 em 10.1.1.101
> CSeq: 802 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Content-Length: 0
>
>
> ---
> Transmitting (NAT) to 10.1.1.101:5060:
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP
> 10.1.1.101:5060;branch=z9hG4bK36da28d784;received=10.1.1.101;rport=5060
> From: "Voiper - 2222" <sip:2222 em asterisk>;tag=5696a690
> To: <sip:3941xxxx em asterisk>;tag=as5771641a
> Call-ID: 628033914eb9a71c7e8a47a42865a0e3 em 10.1.1.101
> CSeq: 802 CANCEL
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Contact: <sip:3941xxxx em 10.1.1.254 <sip%3A3941xxxx em 10.1.1.254>>
> Content-Length: 0
>
>
> ---
> == Spawn extension (macro-saida_azzu, s, 15) exited non-zero on
> 'SIP/22-0817f268' in macro 'saida_azzu'
> == Spawn extension (macro-saida_azzu, s, 15) exited non-zero on
> 'SIP/22-0817f268'
> Retransmitting #1 (NAT) to 10.1.1.101:5060:
> SIP/2.0 487 Request Terminated
> Via: SIP/2.0/UDP
> 10.1.1.101:5060;branch=z9hG4bK36da28d784;received=10.1.1.101;rport=5060
> From: "Voiper - 2222" <sip:2222 em asterisk>;tag=5696a690
> To: <sip:3941xxxx em asterisk>;tag=as5771641a
> Call-ID: 628033914eb9a71c7e8a47a42865a0e3 em 10.1.1.101
> CSeq: 802 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Content-Length: 0
>
>
> ---
> Retransmitting #2 (NAT) to 10.1.1.101:5060:
> SIP/2.0 487 Request Terminated
> Via: SIP/2.0/UDP
> 10.1.1.101:5060;branch=z9hG4bK36da28d784;received=10.1.1.101;rport=5060
> From: "Voiper - 2222" <sip:2222 em asterisk>;tag=5696a690
> To: <sip:3941xxxx em asterisk>;tag=as5771641a
> Call-ID: 628033914eb9a71c7e8a47a42865a0e3 em 10.1.1.101
> CSeq: 802 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Content-Length: 0
>
>
> ---
> testador*CLI>
> <-- SIP read from 10.1.1.101:5060:
> ACK sip:3941xxxx em 10.1.1.254 <sip%3A3941xxxx em 10.1.1.254> SIP/2.0
> Via: SIP/2.0/UDP 10.1.1.101:5060;branch=z9hG4bK36da28d784
> From: "Voiper - 2222" <sip:2222 em asterisk>;tag=5696a690
> To: <sip:3941xxxx em asterisk>;tag=as5771641a
> Call-ID: 628033914eb9a71c7e8a47a42865a0e3 em 10.1.1.101
> Contact: <sip:2222 em 10.1.1.101:5060>
> CSeq: 802 ACK
> Content-Length: 0
>
> --- (8 headers 0 lines) ---
> Destroying call '628033914eb9a71c7e8a47a42865a0e3 em 10.1.1.101'
> testador*CLI>
> <-- SIP read from 10.1.1.101:5060:
> ACK sip:3941xxxx em asterisk SIP/2.0
> Via: SIP/2.0/UDP 10.1.1.101:5060;branch=z9hG4bK36da28d784
> From: "Voiper - 2222" <sip:2222 em asterisk>;tag=5696a690
> To: <sip:3941xxxx em asterisk>;tag=as5771641a
> Call-ID: 628033914eb9a71c7e8a47a42865a0e3 em 10.1.1.101
> CSeq: 802 ACK
> Content-Length: 0
>
> --- (7 headers 0 lines) ---
> Transmitting (NAT) to 10.1.1.101:5060:
> SIP/2.0 481 Call leg/transaction does not exist
> Via: SIP/2.0/UDP
> 10.1.1.101:5060;branch=z9hG4bK36da28d784;received=10.1.1.101
> From: "Voiper - 2222" <sip:2222 em asterisk>;tag=5696a690
> To: <sip:3941xxxx em asterisk>;tag=as5771641a
> Call-ID: 628033914eb9a71c7e8a47a42865a0e3 em 10.1.1.101
> CSeq: 802 ACK
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Content-Length: 0
>
>
> ---
> testador*CLI>
> <-- SIP read from 10.1.1.101:5060:
> ACK sip:3941xxxx em asterisk SIP/2.0
> Via: SIP/2.0/UDP 10.1.1.101:5060;branch=z9hG4bK36da28d784
> From: "Voiper - 2222" <sip:2222 em asterisk>;tag=5696a690
> To: <sip:3941xxxx em asterisk>;tag=as5771641a
> Call-ID: 628033914eb9a71c7e8a47a42865a0e3 em 10.1.1.101
> CSeq: 802 ACK
> Content-Length: 0
>
> --- (7 headers 0 lines) ---
> Transmitting (NAT) to 10.1.1.101:5060:
> SIP/2.0 481 Call leg/transaction does not exist
> Via: SIP/2.0/UDP
> 10.1.1.101:5060;branch=z9hG4bK36da28d784;received=10.1.1.101
> From: "Voiper - 2222" <sip:2222 em asterisk>;tag=5696a690
> To: <sip:3941xxxx em asterisk>;tag=as5771641a
> Call-ID: 628033914eb9a71c7e8a47a42865a0e3 em 10.1.1.101
> CSeq: 802 ACK
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Content-Length: 0
>
>
> ---
> testador*CLI>
> <-- SIP read from 10.1.1.101:5060:
> Via: SIP/2.0/UDP 10.1.1.101:5060;branch=z9hG4bK36da28d784
> From: "Voiper - 2222" <sip:2222 em asterisk>;tag=5696a690
> To: <sip:3941xxxx em asterisk>;tag=as5771641a
> Call-ID: 628033914eb9a71c7e8a47a42865a0e3 em 10.1.1.101
> CSeq: 802 ACK
> Content-Length: 0
>
> --- (6 headers 0 lines) ---
> Mar 18 15:39:31 NOTICE[8300]: chan_sip.c:3989 copy_via_headers: No
> header field 'Via' present to copy
> Transmitting (NAT) to 10.1.1.101:5060:
> SIP/2.0 481 Call leg/transaction does not exist
> From: "Voiper - 2222" <sip:2222 em asterisk>;tag=5696a690
> To: <sip:3941xxxx em asterisk>;tag=as5771641a
> Call-ID: 628033914eb9a71c7e8a47a42865a0e3 em 10.1.1.101
> CSeq: 802 ACK
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Content-Length: 0
>
>
> ---
> testador*CLI>
> <-- SIP read from 10.1.1.101:5060:
> Via: SIP/2.0/UDP 10.1.1.101:5060;branch=z9hG4bK36da28d784
> From: "Voiper - 2222" <sip:2222 em asterisk>;tag=5696a690
> To: <sip:3941xxxx em asterisk>;tag=as5771641a
> Call-ID: 628033914eb9a71c7e8a47a42865a0e3 em 10.1.1.101
> CSeq: 802 ACK
> Content-Length: 0
>
> --- (6 headers 0 lines) ---
> Mar 18 15:39:31 NOTICE[8300]: chan_sip.c:3989 copy_via_headers: No
> header field 'Via' present to copy
> Transmitting (NAT) to 10.1.1.101:5060:
> SIP/2.0 481 Call leg/transaction does not exist
> From: "Voiper - 2222" <sip:2222 em asterisk>;tag=5696a690
> To: <sip:3941xxxx em asterisk>;tag=as5771641a
> Call-ID: 628033914eb9a71c7e8a47a42865a0e3 em 10.1.1.101
> CSeq: 802 ACK
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Content-Length: 0
>
>
> ---
> testador*CLI>
> <-- SIP read from 10.1.1.101:5060:
> Via: SIP/2.0/UDP :5060
> From: "Voiper - 2222" <sip:2222 em asterisk>;tag=5696a690
> To: <sip:3941xxxx em asterisk>;tag=as5771641a
> Call-ID: 628033914eb9a71c7e8a47a42865a0e3 em 10.1.1.101
> CSeq: 802 ACK
> Content-Length: 0
>
> --- (6 headers 0 lines) ---
> Mar 18 15:39:31 NOTICE[8300]: chan_sip.c:3989 copy_via_headers: No
> header field 'Via' present to copy
> Transmitting (NAT) to 10.1.1.101:5060:
> SIP/2.0 481 Call leg/transaction does not exist
> From: "Voiper - 2222" <sip:2222 em asterisk>;tag=5696a690
> To: <sip:3941xxxx em asterisk>;tag=as5771641a
> Call-ID: 628033914eb9a71c7e8a47a42865a0e3 em 10.1.1.101
> CSeq: 802 ACK
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Content-Length: 0
>
>
> ---
> testador*CLI>
> <-- SIP read from 10.1.1.101:5060:
> Via: SIP/2.0/UDP :5060
> From: "Voiper - 2222" <sip:2222 em asterisk>;tag=5696a690
> To: <sip:3941xxxx em asterisk>;tag=as5771641a
> Call-ID: 628033914eb9a71c7e8a47a42865a0e3 em 10.1.1.101
> CSeq: 802 ACK
> Content-Length: 0
>
> --- (6 headers 0 lines) ---
> Mar 18 15:39:32 NOTICE[8300]: chan_sip.c:3989 copy_via_headers: No
> header field 'Via' present to copy
> Transmitting (NAT) to 10.1.1.101:5060:
> SIP/2.0 481 Call leg/transaction does not exist
> From: "Voiper - 2222" <sip:2222 em asterisk>;tag=5696a690
> To: <sip:3941xxxx em asterisk>;tag=as5771641a
> Call-ID: 628033914eb9a71c7e8a47a42865a0e3 em 10.1.1.101
> CSeq: 802 ACK
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Content-Length: 0
>
> _______________________________________________
> Openmoko Freerunner, primeiro telefone open source, disponível no Brasil
> rodando o Android da Google.
> http://www.neodroid.com
>
> Compre uma camiseta da AsteriskBrasil.org!
> http://www.voipmania.com.br
>
> Acesse o canal IRC de discussão sobre Asterisk em Português Brasileiro na
> rede Freenode.net: #asterisk-br
> _______________________________________________
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> AsteriskBrasil em listas.asteriskbrasil.org
> http://listas.asteriskbrasil.org/mailman/listinfo/asteriskbrasil
>
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