[AsteriskBrasil] [Pacotes SIP] Problema com Telefone Voiper da Intelbrás
Junior Polegato - Asterisk
asterisk em juniorpolegato.com.br
Quarta Março 18 16:06:25 BRT 2009
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>
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>;tag=as027fc344
To: <sip:2222 em 10.1.1.101:5060>
Contact: <sip:asterisk 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>;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>
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/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
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