[AsteriskBrasil] Problemas com T.38
Daviramos Roussenq Fortunato
daviramosrf em gmail.com
Sexta Maio 29 18:17:25 BRT 2009
Boa Tarde Lista.
Estou com problemas na tramissão de fax utilizando T.38.
Meu cenario é o seguinte:
Asterisk 1.6.0.5
2 ATAS 2210 T da Intelbras.
ReceiveFAX no proprio asterisk.
Não consigo passa fax quando é de um ATA para outro utlizando o Asterisk
no meio, se faço direto entre os ATA funciona perfeitamente, se passo de um
ATA para o ReceiveFAX do Asterisk funciona perfeito, mas se tento passar
entre dois RAMAIS utilizando o mesmo ATA, não funciona nunca.
rtp.conf:
[general]
rtpstart=17000
rtpend=33000
udptl.conf:
[general]
udptlstart=4000
udptlend=4999
T38FaxUdpEC = t38UDPRedundancy
T38FaxMaxDatagram = 400
udptlfecentries = 3
udptlfecspan = 3
sip.conf:
t38pt_udptl = yes
t38pt_rtp=no
t38pt_tcp=no
Fiz um tcpdump e quando logo após ouvir dar o sinal de fax para envio, o
trafego de rede rtp para e só ao final da ligação no timeout mostra mais
alguns dados do protocolo SIP e a ligação cai.
Exemplo de quando é dado o sinal de fax:
18:05:28.931701 IP XX.XX.XX.67.1024 > XX.XX.XX.66.sip: SIP, length: 439
18:05:29.047186 IP XX.XX.XX.67.1024 > XX.XX.XX.66.sip: SIP, length: 573
18:05:29.163231 IP XX.XX.XX.67.1024 > XX.XX.XX.66.sip: SIP, length: 439
18:05:31.336965 IP XX.XX.XX.67 > XX.XX.XX.66: ICMP XX.XX.XX.67 udp port
17003 unreachable, length 36
18:05:36.339933 IP XX.XX.XX.67 > XX.XX.XX.66: ICMP XX.XX.XX.67 udp port
17003 unreachable, length 36
18:05:41.338790 IP XX.XX.XX.67 > XX.XX.XX.66: ICMP XX.XX.XX.67 udp port
17003 unreachable, length 36
Mas quando uso ATA para o ReceiveFAX, o trafego rtp fica constante até
terminar a passagem do fax.
Segue sip debug peer do momento que o fax não passa:
<--- Transmitting (NAT) to XX.XX.XX.67:1024 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.2.100:5060
;branch=z9hG4bKdN0f2-0Uic90FJs;received=XX.XX.XX.67
From: 2005618<sip:2005618 em XX.XX.XX.66>;tag=74tf2-7Q4xE0
To: sip:2007 em XX.XX.XX.66;tag=as46031e07
Call-ID: CcD6S0-VFf0Z8f2 em XX.XX.XX.66
CSeq: 51 INVITE
User-Agent: Asterisk PBX 1.6.0.5
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:2007 em XX.XX.XX.66>
Content-Length: 0
<------------>
Audio is at XX.XX.XX.66 port 30206
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
engeplus*CLI>
<--- Transmitting (NAT) to XX.XX.XX.67:1024 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.2.100:5060
;branch=z9hG4bKdN0f2-0Uic90FJs;received=XX.XX.XX.67
From: 2005618<sip:2005618 em XX.XX.XX.66>;tag=74tf2-7Q4xE0
To: sip:2007 em XX.XX.XX.66;tag=as46031e07
Call-ID: CcD6S0-VFf0Z8f2 em XX.XX.XX.66
CSeq: 51 INVITE
User-Agent: Asterisk PBX 1.6.0.5
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:2007 em XX.XX.XX.66>
Content-Type: application/sdp
Content-Length: 283
v=0
o=root 577489170 577489170 IN IP4 XX.XX.XX.66
s=Asterisk PBX 1.6.0.5
c=IN IP4 XX.XX.XX.66
t=0 0
m=audio 30206 RTP/AVP 18 96
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------>
-- SIP/2007618-b7cb7cf8 is ringing
-- SIP/2007618-b7cb7cf8 answered SIP/2005618-08bd3e50
Audio is at XX.XX.XX.66 port 30206
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Reliably Transmitting (NAT) to XX.XX.XX.67:1024 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.100:5060
;branch=z9hG4bKdN0f2-0Uic90FJs;received=XX.XX.XX.67
From: 2005618<sip:2005618 em XX.XX.XX.66>;tag=74tf2-7Q4xE0
To: sip:2007 em XX.XX.XX.66;tag=as46031e07
Call-ID: CcD6S0-VFf0Z8f2 em XX.XX.XX.66
CSeq: 51 INVITE
User-Agent: Asterisk PBX 1.6.0.5
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:2007 em XX.XX.XX.66>
Content-Type: application/sdp
Content-Length: 283
v=0
o=root 577489170 577489171 IN IP4 XX.XX.XX.66
s=Asterisk PBX 1.6.0.5
c=IN IP4 XX.XX.XX.66
t=0 0
m=audio 30206 RTP/AVP 18 96
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------>
engeplus*CLI>
<--- SIP read from UDP://XX.XX.XX.67:1024 --->
ACK sip:2007 em XX.XX.XX.66:5060 SIP/2.0
From: 2005618<sip:2005618 em XX.XX.XX.66>;tag=74tf2-7Q4xE0
To: sip:2007 em XX.XX.XX.66;tag=as46031e07
Call-ID: CcD6S0-VFf0Z8f2 em XX.XX.XX.66
CSeq: 51 ACK
Via: SIP/2.0/UDP 192.168.2.100:5060;branch=z9hG4bKdN0f2-iNjO1len
Contact: 2005618<sip:2005618 em 192.168.2.100:5060>
Max-Forwards: 70
User-Agent: INTELBRAS ATA GKM2210T - Nov 19 2008
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
set_destination: Parsing <sip:2005618 em 192.168.2.100:5060> for address/port
to send to
set_destination: set destination to 192.168.2.100, port 5060
Reliably Transmitting (NAT) to XX.XX.XX.67:1024:
INVITE sip:2005618 em 192.168.2.100:5060 SIP/2.0
Via: SIP/2.0/UDP XX.XX.XX.66:5060;branch=z9hG4bK3380e2dc;rport
Max-Forwards: 70
From: sip:2007 em XX.XX.XX.66;tag=as46031e07
To: 2005618<sip:2005618 em XX.XX.XX.66>;tag=74tf2-7Q4xE0
Contact: <sip:2007 em XX.XX.XX.66>
Call-ID: CcD6S0-VFf0Z8f2 em XX.XX.XX.66
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.0.5
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 365
v=0
o=root 577489170 577489172 IN IP4 XX.XX.XX.66
s=Asterisk PBX 1.6.0.5
c=IN IP4 XX.XX.XX.66
t=0 0
m=image 4729 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:400
a=T38FaxMaxDatagram:400
a=T38FaxUdpEC:t38UDPFEC
---
engeplus*CLI>
<--- SIP read from UDP://XX.XX.XX.67:1024 --->
SIP/2.0 100 Trying
From: sip:2007 em XX.XX.XX.66;tag=as46031e07
To: 2005618<sip:2005618 em XX.XX.XX.66>;tag=74tf2-7Q4xE0
Call-ID: CcD6S0-VFf0Z8f2 em XX.XX.XX.66
CSeq: 102 INVITE
Via: SIP/2.0/UDP XX.XX.XX.66:5060;branch=z9hG4bK3380e2dc
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
<--- SIP read from UDP://XX.XX.XX.67:1024 --->
SIP/2.0 200 OK
From: sip:2007 em XX.XX.XX.66;tag=as46031e07
To: 2005618<sip:2005618 em XX.XX.XX.66>;tag=74tf2-7Q4xE0
Call-ID: CcD6S0-VFf0Z8f2 em XX.XX.XX.66
CSeq: 102 INVITE
Via: SIP/2.0/UDP XX.XX.XX.66:5060;branch=z9hG4bK3380e2dc
Contact: 2005618<sip:2005618 em 192.168.2.100:5060>
User-Agent: INTELBRAS ATA GKM2210T - Nov 19 2008
Allow: INVITE,BYE,CANCEL,OPTIONS,PRACK,NOTIFY,UPDATE,REFER
Supported: timer,replaces
Content-Type: application/sdp
Content-Length: 243
v=0
o=2005618 207176 2 IN IP4 192.168.2.100
s=-
c=IN IP4 192.168.2.100
t=0 0
m=image 17002 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxMaxBuffer:400
a=T38FaxUdpEc:t38UDPRedundancy
a=T38FaxRateManagement:transferredTCF
<------------->
--- (12 headers 11 lines) ---
Got T.38 offer in SDP in dialog CcD6S0-VFf0Z8f2 em XX.XX.XX.66
Got T.38 Re-invite without audio. Keeping RTP active during T.38 session.
Callid CcD6S0-VFf0Z8f2 em XX.XX.XX.66
Capabilities: us - 0x102 (gsm|g729), peer - audio=0x0 (nothing)/video=0x0
(nothing)/text=0x0 (nothing), combined - 0x0 (nothing)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0
(nothing), combined - 0x0 (nothing)
set_destination: Parsing <sip:2005618 em 192.168.2.100:5060> for address/port
to send to
set_destination: set destination to 192.168.2.100, port 5060
Transmitting (NAT) to XX.XX.XX.67:1024:
ACK sip:2005618 em 192.168.2.100:5060 SIP/2.0
Via: SIP/2.0/UDP XX.XX.XX.66:5060;branch=z9hG4bK732ec166;rport
Max-Forwards: 70
From: sip:2007 em XX.XX.XX.66;tag=as46031e07
To: 2005618<sip:2005618 em XX.XX.XX.66>;tag=74tf2-7Q4xE0
Contact: <sip:2007 em XX.XX.XX.66>
Call-ID: CcD6S0-VFf0Z8f2 em XX.XX.XX.66
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.0.5
Content-Length: 0
---
engeplus*CLI>
Alguém sabe tem ideia do que fazer? Já testei varias configurações
possiveis.
-------------- Próxima Parte ----------
Um anexo em HTML foi limpo...
URL: http://listas.asteriskbrasil.org/pipermail/asteriskbrasil/attachments/20090529/8be5d1cc/attachment-0001.htm
Mais detalhes sobre a lista de discussão AsteriskBrasil