[AsteriskBrasil] RES: X100p Clone continua ...
Emanuel dos Reis Rodrigues
emanueldosreis em gmail.com
Terça Novembro 11 21:47:16 BRST 2008
Giancarlo Rubio wrote:
> 2008/11/11 Emanuel dos Reis Rodrigues <emanueldosreis em gmail.com>:
>
> Está parecendo que sua placa está queimada, mal contato ou algo do tipo.
>
> Tente debugar a chamada sip para saber se seu asterisk se comunica
> realmente com seu ata. No cli do seu asterisk digite
>
> sip set debug peer 101
>
> A partir dai vc pode elimar o problema da rede/ata ou do seu asterisk/placa.
>
>
>
>> Olha que estranho, fiz uma ligação para meu telefone, esperei muito
>> tempo e desliguei ... depois de uns minutos do nada o telefone tocou ....
>>
>>
>> o que pode ser ? ligação com 3 minutos de atraso ...
>>
>>
>> --
>> Emanuel dos Reis Rodrigues
>> Senior Level Linux Professional (LPIC-3)
>> emanueldosreis(No*SpAm)gmail.com
>> +55 95 8112-9628
>>
>>
>>
>>
>> _______________________________________________
>> 2 a 4 de Dezembro - IPComm 2008
>> Com presença de engenheiros da Digium e Jon "maddog" Hall.
>> http://www.ipcomm2008.com.br
>>
>> 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
>>
>>
>
>
>
>
olha só:
Estou usando uma ATA GKM2000 grandstream com um telefone normal ligado a
ele, que é o ramail SIP 101 ... mas também fiz testes com o softphone ...
o ip do ATA é 10.30.1.11 e do asterisk 10.30.1.6
O modem, ligo a máquina e paro ela no grub, depois conecto a linha e um
telefone no modem e consigo fazer ligações normalmente ...
Fiz o debug que me pediu:
Disquei de um telefone para o meu ligado no modem:
<--- SIP read from UDP://10.30.1.11:5062 --->
<------------->
[Nov 11 19:37:42] ERROR[3267]: callerid.c:562 callerid_feed: No start
bit found in fsk data.
[Nov 11 19:37:42] WARNING[3267]: chan_dahdi.c:7154 ss_thread: CallerID
feed failed: Success
[Nov 11 19:37:42] WARNING[3267]: chan_dahdi.c:7258 ss_thread: CallerID
returned with error on channel 'DAHDI/1-1'
Audio is at 10.30.1.6 port 11666
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.30.1.11:5062:
INVITE sip:101 em 10.30.1.11:5062;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.30.1.6:5060;branch=z9hG4bK16b494f4;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk em 10.30.1.6>;tag=as541c141d
To: <sip:101 em 10.30.1.11:5062;user=phone>
Contact: <sip:asterisk em 10.30.1.6>
Call-ID: 6006c14002ff0b07268a6f8c33ad93e8 em 10.30.1.6
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.0.1
Date: Tue, 11 Nov 2008 23:37:42 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 304
v=0
o=root 2039819054 2039819054 IN IP4 10.30.1.6
s=Asterisk PBX 1.6.0.1
c=IN IP4 10.30.1.6
t=0 0
m=audio 11666 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
asterisk*CLI>
<--- SIP read from UDP://10.30.1.11:5062 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.30.1.6:5060;branch=z9hG4bK16b494f4;rport
From: "asterisk" <sip:asterisk em 10.30.1.6>;tag=as541c141d
To: <sip:101 em 10.30.1.11:5062;user=phone>
Call-ID: 6006c14002ff0b07268a6f8c33ad93e8 em 10.30.1.6
CSeq: 102 INVITE
User-Agent: Grandstream HT386 1.0.3.77 FXS1
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
asterisk*CLI>
<--- SIP read from UDP://10.30.1.11:5062 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.30.1.6:5060;branch=z9hG4bK16b494f4;rport
From: "asterisk" <sip:asterisk em 10.30.1.6>;tag=as541c141d
To: <sip:101 em 10.30.1.11:5062;user=phone>;tag=358898645d36cc16
Call-ID: 6006c14002ff0b07268a6f8c33ad93e8 em 10.30.1.6
CSeq: 102 INVITE
User-Agent: Grandstream HT386 1.0.3.77 FXS1
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
Scheduling destruction of SIP dialog
'6006c14002ff0b07268a6f8c33ad93e8 em 10.30.1.6' in 32000 ms (Method: INVITE)
Reliably Transmitting (no NAT) to 10.30.1.11:5062:
CANCEL sip:101 em 10.30.1.11:5062;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.30.1.6:5060;branch=z9hG4bK16b494f4;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk em 10.30.1.6>;tag=as541c141d
To: <sip:101 em 10.30.1.11:5062;user=phone>
Call-ID: 6006c14002ff0b07268a6f8c33ad93e8 em 10.30.1.6
CSeq: 102 CANCEL
User-Agent: Asterisk PBX 1.6.0.1
Content-Length: 0
---
Scheduling destruction of SIP dialog
'6006c14002ff0b07268a6f8c33ad93e8 em 10.30.1.6' in 32000 ms (Method: INVITE)
asterisk*CLI>
<--- SIP read from UDP://10.30.1.11:5062 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.30.1.6:5060;branch=z9hG4bK16b494f4;rport
From: "asterisk" <sip:asterisk em 10.30.1.6>;tag=as541c141d
To: <sip:101 em 10.30.1.11:5062;user=phone>;tag=358898645d36cc16
Call-ID: 6006c14002ff0b07268a6f8c33ad93e8 em 10.30.1.6
CSeq: 102 CANCEL
User-Agent: Grandstream HT386 1.0.3.77 FXS1
Contact: <sip:101 em 10.30.1.11:5062;user=phone>
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Supported: replaces
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
asterisk*CLI>
<--- SIP read from UDP://10.30.1.11:5062 --->
SIP/2.0 487 Request Cancelled
Via: SIP/2.0/UDP 10.30.1.6:5060;branch=z9hG4bK16b494f4;rport
From: "asterisk" <sip:asterisk em 10.30.1.6>;tag=as541c141d
To: <sip:101 em 10.30.1.11:5062;user=phone>;tag=358898645d36cc16
Call-ID: 6006c14002ff0b07268a6f8c33ad93e8 em 10.30.1.6
CSeq: 102 INVITE
User-Agent: Grandstream HT386 1.0.3.77 FXS1
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
Transmitting (no NAT) to 10.30.1.11:5062:
ACK sip:101 em 10.30.1.11:5062;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.30.1.6:5060;branch=z9hG4bK16b494f4;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk em 10.30.1.6>;tag=as541c141d
To: <sip:101 em 10.30.1.11:5062;user=phone>;tag=358898645d36cc16
Contact: <sip:asterisk em 10.30.1.6>
Call-ID: 6006c14002ff0b07268a6f8c33ad93e8 em 10.30.1.6
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.0.1
Content-Length: 0
---
Really destroying SIP dialog
'6006c14002ff0b07268a6f8c33ad93e8 em 10.30.1.6' Method: INVITE
asterisk*CLI>
<--- SIP read from UDP://10.30.1.11:5062 --->
<------------->
[Nov 11 19:38:03] NOTICE[3268]: chan_dahdi.c:7114 ss_thread: Got event
18 (Ring Begin)...
[Nov 11 19:38:03] NOTICE[3268]: chan_dahdi.c:7114 ss_thread: Got event 2
(Ring/Answered)...
[Nov 11 19:38:03] NOTICE[3268]: chan_dahdi.c:7114 ss_thread: Got event
18 (Ring Begin)...
Audio is at 10.30.1.6 port 19950
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.30.1.11:5062:
INVITE sip:101 em 10.30.1.11:5062;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.30.1.6:5060;branch=z9hG4bK6be47d1c;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk em 10.30.1.6>;tag=as4b92e8e8
To: <sip:101 em 10.30.1.11:5062;user=phone>
Contact: <sip:asterisk em 10.30.1.6>
Call-ID: 21ff830105ba4d5569e195965a3995ea em 10.30.1.6
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.0.1
Date: Tue, 11 Nov 2008 23:38:03 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 304
v=0
o=root 1337566865 1337566865 IN IP4 10.30.1.6
s=Asterisk PBX 1.6.0.1
c=IN IP4 10.30.1.6
t=0 0
m=audio 19950 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
asterisk*CLI>
<--- SIP read from UDP://10.30.1.11:5062 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.30.1.6:5060;branch=z9hG4bK6be47d1c;rport
From: "asterisk" <sip:asterisk em 10.30.1.6>;tag=as4b92e8e8
To: <sip:101 em 10.30.1.11:5062;user=phone>
Call-ID: 21ff830105ba4d5569e195965a3995ea em 10.30.1.6
CSeq: 102 INVITE
User-Agent: Grandstream HT386 1.0.3.77 FXS1
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
<--- SIP read from UDP://10.30.1.11:5062 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.30.1.6:5060;branch=z9hG4bK6be47d1c;rport
From: "asterisk" <sip:asterisk em 10.30.1.6>;tag=as4b92e8e8
To: <sip:101 em 10.30.1.11:5062;user=phone>;tag=d2ba412a375537cc
Call-ID: 21ff830105ba4d5569e195965a3995ea em 10.30.1.6
CSeq: 102 INVITE
User-Agent: Grandstream HT386 1.0.3.77 FXS1
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
Scheduling destruction of SIP dialog
'21ff830105ba4d5569e195965a3995ea em 10.30.1.6' in 32000 ms (Method: INVITE)
Reliably Transmitting (no NAT) to 10.30.1.11:5062:
CANCEL sip:101 em 10.30.1.11:5062;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.30.1.6:5060;branch=z9hG4bK6be47d1c;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk em 10.30.1.6>;tag=as4b92e8e8
To: <sip:101 em 10.30.1.11:5062;user=phone>
Call-ID: 21ff830105ba4d5569e195965a3995ea em 10.30.1.6
CSeq: 102 CANCEL
User-Agent: Asterisk PBX 1.6.0.1
Content-Length: 0
---
Scheduling destruction of SIP dialog
'21ff830105ba4d5569e195965a3995ea em 10.30.1.6' in 32000 ms (Method: INVITE)
asterisk*CLI>
<--- SIP read from UDP://10.30.1.11:5062 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.30.1.6:5060;branch=z9hG4bK6be47d1c;rport
From: "asterisk" <sip:asterisk em 10.30.1.6>;tag=as4b92e8e8
To: <sip:101 em 10.30.1.11:5062;user=phone>;tag=d2ba412a375537cc
Call-ID: 21ff830105ba4d5569e195965a3995ea em 10.30.1.6
CSeq: 102 CANCEL
User-Agent: Grandstream HT386 1.0.3.77 FXS1
Contact: <sip:101 em 10.30.1.11:5062;user=phone>
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Supported: replaces
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
asterisk*CLI>
<--- SIP read from UDP://10.30.1.11:5062 --->
SIP/2.0 487 Request Cancelled
Via: SIP/2.0/UDP 10.30.1.6:5060;branch=z9hG4bK6be47d1c;rport
From: "asterisk" <sip:asterisk em 10.30.1.6>;tag=as4b92e8e8
To: <sip:101 em 10.30.1.11:5062;user=phone>;tag=d2ba412a375537cc
Call-ID: 21ff830105ba4d5569e195965a3995ea em 10.30.1.6
CSeq: 102 INVITE
User-Agent: Grandstream HT386 1.0.3.77 FXS1
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
Transmitting (no NAT) to 10.30.1.11:5062:
ACK sip:101 em 10.30.1.11:5062;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.30.1.6:5060;branch=z9hG4bK6be47d1c;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk em 10.30.1.6>;tag=as4b92e8e8
To: <sip:101 em 10.30.1.11:5062;user=phone>;tag=d2ba412a375537cc
Contact: <sip:asterisk em 10.30.1.6>
Call-ID: 21ff830105ba4d5569e195965a3995ea em 10.30.1.6
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.0.1
Content-Length: 0
---
Really destroying SIP dialog
'21ff830105ba4d5569e195965a3995ea em 10.30.1.6' Method: INVITE
asterisk*CLI>
<--- SIP read from UDP://10.30.1.11:5062 --->
<------------->
[Nov 11 19:38:21] NOTICE[3269]: chan_dahdi.c:7114 ss_thread: Got event
18 (Ring Begin)...
Audio is at 10.30.1.6 port 11086
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.30.1.11:5062:
INVITE sip:101 em 10.30.1.11:5062;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.30.1.6:5060;branch=z9hG4bK1a888e62;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk em 10.30.1.6>;tag=as2c6e6074
To: <sip:101 em 10.30.1.11:5062;user=phone>
Contact: <sip:asterisk em 10.30.1.6>
Call-ID: 304a7a277f936b2f07d566db31a3c48b em 10.30.1.6
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.0.1
Date: Tue, 11 Nov 2008 23:38:21 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 302
v=0
o=root 742182519 742182519 IN IP4 10.30.1.6
s=Asterisk PBX 1.6.0.1
c=IN IP4 10.30.1.6
t=0 0
m=audio 11086 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
asterisk*CLI>
<--- SIP read from UDP://10.30.1.11:5062 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.30.1.6:5060;branch=z9hG4bK1a888e62;rport
From: "asterisk" <sip:asterisk em 10.30.1.6>;tag=as2c6e6074
To: <sip:101 em 10.30.1.11:5062;user=phone>
Call-ID: 304a7a277f936b2f07d566db31a3c48b em 10.30.1.6
CSeq: 102 INVITE
User-Agent: Grandstream HT386 1.0.3.77 FXS1
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
<--- SIP read from UDP://10.30.1.11:5062 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.30.1.6:5060;branch=z9hG4bK1a888e62;rport
From: "asterisk" <sip:asterisk em 10.30.1.6>;tag=as2c6e6074
To: <sip:101 em 10.30.1.11:5062;user=phone>;tag=76f5b74c5a4cb63d
Call-ID: 304a7a277f936b2f07d566db31a3c48b em 10.30.1.6
CSeq: 102 INVITE
User-Agent: Grandstream HT386 1.0.3.77 FXS1
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
Scheduling destruction of SIP dialog
'304a7a277f936b2f07d566db31a3c48b em 10.30.1.6' in 32000 ms (Method: INVITE)
Reliably Transmitting (no NAT) to 10.30.1.11:5062:
CANCEL sip:101 em 10.30.1.11:5062;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.30.1.6:5060;branch=z9hG4bK1a888e62;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk em 10.30.1.6>;tag=as2c6e6074
To: <sip:101 em 10.30.1.11:5062;user=phone>
Call-ID: 304a7a277f936b2f07d566db31a3c48b em 10.30.1.6
CSeq: 102 CANCEL
User-Agent: Asterisk PBX 1.6.0.1
Content-Length: 0
---
Scheduling destruction of SIP dialog
'304a7a277f936b2f07d566db31a3c48b em 10.30.1.6' in 32000 ms (Method: INVITE)
asterisk*CLI>
<--- SIP read from UDP://10.30.1.11:5062 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.30.1.6:5060;branch=z9hG4bK1a888e62;rport
From: "asterisk" <sip:asterisk em 10.30.1.6>;tag=as2c6e6074
To: <sip:101 em 10.30.1.11:5062;user=phone>;tag=76f5b74c5a4cb63d
Call-ID: 304a7a277f936b2f07d566db31a3c48b em 10.30.1.6
CSeq: 102 CANCEL
User-Agent: Grandstream HT386 1.0.3.77 FXS1
Contact: <sip:101 em 10.30.1.11:5062;user=phone>
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Supported: replaces
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
asterisk*CLI>
<--- SIP read from UDP://10.30.1.11:5062 --->
SIP/2.0 487 Request Cancelled
Via: SIP/2.0/UDP 10.30.1.6:5060;branch=z9hG4bK1a888e62;rport
From: "asterisk" <sip:asterisk em 10.30.1.6>;tag=as2c6e6074
To: <sip:101 em 10.30.1.11:5062;user=phone>;tag=76f5b74c5a4cb63d
Call-ID: 304a7a277f936b2f07d566db31a3c48b em 10.30.1.6
CSeq: 102 INVITE
User-Agent: Grandstream HT386 1.0.3.77 FXS1
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
Transmitting (no NAT) to 10.30.1.11:5062:
ACK sip:101 em 10.30.1.11:5062;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.30.1.6:5060;branch=z9hG4bK1a888e62;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk em 10.30.1.6>;tag=as2c6e6074
To: <sip:101 em 10.30.1.11:5062;user=phone>;tag=76f5b74c5a4cb63d
Contact: <sip:asterisk em 10.30.1.6>
Call-ID: 304a7a277f936b2f07d566db31a3c48b em 10.30.1.6
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.0.1
Content-Length: 0
---
Really destroying SIP dialog
'304a7a277f936b2f07d566db31a3c48b em 10.30.1.6' Method: INVITE
asterisk*CLI>
<--- SIP read from UDP://10.30.1.11:5062 --->
Valeu ....
Emanuel
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