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Marcus Vinicius

2009-Nov-14 15:20 UTC

[asterisk-users] Asterisk with T38 Fax

Hi, 

I'm trying to send faxes using Asterisk 1.4 and T38 with sip but Asterisk
rejects the t38.

Anybody know if is possible to transmit t38 fax with Asterisk 1.4?

following settings:

--- sip.conf ---

[general]
bindport=5060
bindaddr=0.0.0.0
disallow=all
allow=ulaw
allow=alaw
context=from-outside
t38pt_udptl=yes


[operator]
qualify=no
nat=yes
host=189.160.126.201
dtmfmode=rfc2833
context=from-outside
type=friend
canreinvite=yes
t38pt_udptl=yes
;t38pt_rtp=no
;t38pt_tcp=no
disallow=all
allow=ulaw
allow=alaw



--- channels/chan_sip.c ---

static int global_t38_capability = T38FAX_VERSION_0 | T38FAX_RATE_2400 |
T38FAX_RATE_4800 | T38FAX_RATE_7200 | T38FAX_RATE_9600;


--- logs ---


logs

[Nov 13 10:21:11] VERBOSE[25087] logger.c:
<--- SIP read from 189.160.126.210:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 189.6.70.47:5060;branch=z9hG4bK6de18e85;rport=5060
Call-ID: 21cdaea43523056c3a09c45b13c9a940 at 189.6.70.47
From: "Teste"<sip:3013 at 189.6.70.47>;tag=as41b028c6
To: <sip:0411331644005 at 189.160.126.210>;tag=66359f37
CSeq: 102 INVITE
Allow:
INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER
Contact: <sip:0411331644005 at 189.160.126.210:5060;user=phone>
Content-Length: 237
Content-Type: application/sdp

v=0
o=HuaweiSoftX3000 31955175 31955175 IN IP4 189.160.126.210
s=Sip Call
c=IN IP4 189.160.126.210
t=0 0
m=audio 13474 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

<------------->
[Nov 13 10:21:11] VERBOSE[25087] logger.c: --- (10 headers 10 lines) ---
[Nov 13 10:21:11] VERBOSE[25087] logger.c: Found RTP audio format 0
[Nov 13 10:21:11] VERBOSE[25087] logger.c: Found RTP audio format 8
[Nov 13 10:21:11] VERBOSE[25087] logger.c: Found RTP audio format 101
[Nov 13 10:21:11] DEBUG[25087] chan_sip.c: Peer doesn't provide T.38 UDPTL
[Nov 13 10:21:11] VERBOSE[25087] logger.c: Peer audio RTP is at port
189.160.126.210:13474
[Nov 13 10:21:11] VERBOSE[25087] logger.c: Found audio description format PCMU
for ID 0
[Nov 13 10:21:11] VERBOSE[25087] logger.c: Found audio description format PCMA
for ID 8
[Nov 13 10:21:11] VERBOSE[25087] logger.c: Found audio description format
telephone-event for ID 101
[Nov 13 10:21:11] VERBOSE[25087] logger.c: Capabilities: us - 0xc (ulaw|alaw),
peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
[Nov 13 10:21:11] VERBOSE[25087] logger.c: Non-codec capabilities (dtmf): us -
0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1
(telephone-event)
[Nov 13 10:21:11] VERBOSE[25087] logger.c: Peer audio RTP is at port
189.160.126.210:13474
[Nov 13 10:21:11] VERBOSE[13464] logger.c:     -- SIP/ctbc-08345a10 is making
progress passing it to IAX2/nmg010-to-nmg005-trunk1-2748


<--- SIP read from 189.160.126.210:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 189.6.70.47:5060;branch=z9hG4bK6de18e85;rport=5060
Call-ID: 21cdaea43523056c3a09c45b13c9a940 at 189.6.70.47
From: "Teste"<sip:3013 at 189.6.70.47>;tag=as41b028c6
To: <sip:0411331644005 at 189.160.126.210>;tag=66359f37
CSeq: 102 INVITE
Contact: <sip:0411331644005 at 189.160.126.210:5060;user=phone>
Content-Length: 237
Content-Type: application/sdp

v=0
o=HuaweiSoftX3000 31955175 31955176 IN IP4 189.160.126.210
s=Sip Call
c=IN IP4 189.160.126.210
t=0 0
m=audio 13474 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

<------------->
[Nov 13 10:21:15] VERBOSE[25087] logger.c: --- (9 headers 10 lines) ---
[Nov 13 10:21:15] VERBOSE[25087] logger.c: Found RTP audio format 0
[Nov 13 10:21:15] VERBOSE[25087] logger.c: Found RTP audio format 8
[Nov 13 10:21:15] VERBOSE[25087] logger.c: Found RTP audio format 101
[Nov 13 10:21:15] DEBUG[25087] chan_sip.c: Peer doesn't provide T.38 UDPTL
[Nov 13 10:21:15] VERBOSE[25087] logger.c: Peer audio RTP is at port
189.160.126.210:13474
[Nov 13 10:21:15] VERBOSE[25087] logger.c: Found audio description format PCMU
for ID 0
[Nov 13 10:21:15] VERBOSE[25087] logger.c: Found audio description format PCMA
for ID 8
[Nov 13 10:21:15] VERBOSE[25087] logger.c: Found audio description format
telephone-event for ID 101
[Nov 13 10:21:15] VERBOSE[25087] logger.c: Capabilities: us - 0xc (ulaw|alaw),
peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
[Nov 13 10:21:15] VERBOSE[25087] logger.c: Non-codec capabilities (dtmf): us -
0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1
(telephone-event)
[Nov 13 10:21:15] VERBOSE[25087] logger.c: Peer audio RTP is at port
189.160.126.210:13474
[Nov 13 10:21:15] VERBOSE[25087] logger.c: list_route: hop:
<sip:0411331644005 at 189.160.126.210:5060;user=phone>
[Nov 13 10:21:15] DEBUG[25087] chan_sip.c: Strict routing enforced for session
21cdaea43523056c3a09c45b13c9a940 at 189.6.70.47
[Nov 13 10:21:15] VERBOSE[25087] logger.c: set_destination: Parsing
<sip:0411331644005 at 189.160.126.210:5060;user=phone> for address/port to
send to
[Nov 13 10:21:15] VERBOSE[25087] logger.c: set_destination: set destination to
189.160.126.210, port 5060
[Nov 13 10:21:15] VERBOSE[25087] logger.c: Transmitting (NAT) to
189.160.126.210:5060:


ACK sip:0411331644005 at 189.160.126.210:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 189.6.70.47:5060;branch=z9hG4bK6618dc53;rport
From: "Teste" <sip:3013 at 189.6.70.47>;tag=as41b028c6
To: <sip:0411331644005 at 189.160.126.210>;tag=66359f37
Contact: <sip:3013 at 189.6.70.47>
Call-ID: 21cdaea43523056c3a09c45b13c9a940 at 189.6.70.47
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
[Nov 13 10:21:15] VERBOSE[13464] logger.c:     -- SIP/ctbc-08345a10 answered
IAX2/nmg010-to-nmg005-trunk1-2748


<--- SIP read from 189.160.126.210:5060 --->
INVITE sip:3013 at 189.6.70.47 SIP/2.0
Via: SIP/2.0/UDP 189.160.126.210:5060;branch=z9hG4bKb9f6b05aa5fb314af51ece37c
Call-ID: 21cdaea43523056c3a09c45b13c9a940 at 189.6.70.47
From: <sip:0411331644005 at 189.160.126.210>;tag=66359f37
To: "Teste"<sip:3013 at 189.6.70.47>;tag=as41b028c6
CSeq: 1 INVITE
Max-Forwards: 70
Contact: <sip:189.160.126.210:5060>
Content-Length: 295
Content-Type: application/sdp

v=0
o=HuaweiSoftX3000 31955175 31955177 IN IP4 189.160.126.210
s=Sip Call
c=IN IP4 189.160.126.210
t=0 0
m=image 13474 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:176
a=T38FaxMaxDatagram:176
a=T38FaxUdpEC:t38UDPRedundancy

<------------->
[Nov 13 10:21:18] VERBOSE[25087] logger.c: --- (10 headers 12 lines) ---
[Nov 13 10:21:18] VERBOSE[25087] logger.c: Sending to 189.160.126.210 : 5060
(NAT)
[Nov 13 10:21:18] VERBOSE[25087] logger.c: Got T.38 offer in SDP in dialog
21cdaea43523056c3a09c45b13c9a940 at 189.6.70.47
[Nov 13 10:21:18] DEBUG[25087] chan_sip.c: Peer T.38 UDPTL is at port
189.160.126.210:13474
[Nov 13 10:21:18] VERBOSE[25087] logger.c: Got T.38 Re-invite without audio.
Keeping RTP active during T.38 session. Callid 21cdaea43523056c3a09c45b13c9a940
at 189.6.70.47
[Nov 13 10:21:18] DEBUG[25087] chan_sip.c: Our T38 capability = (16208), peer
T38 capability (3872), joint T38 capability (3872)

[Nov 13 10:21:18] VERBOSE[25087] logger.c: Capabilities: us - 0xc (ulaw|alaw),
peer - audio=0x0 (nothing)/video=0x0 (nothing), combined - 0x0 (nothing)
[Nov 13 10:21:18] VERBOSE[25087] logger.c: Non-codec capabilities (dtmf): us -
0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
[Nov 13 10:21:18] VERBOSE[25087] logger.c:
<--- Transmitting (NAT) to 189.160.126.210:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
189.160.126.210:5060;branch=z9hG4bKb9f6b05aa5fb314af51ece37c;received=189.160.126.210
From: <sip:0411331644005 at 189.160.126.210>;tag=66359f37
To: "Teste"<sip:3013 at 189.6.70.47>;tag=as41b028c6
Call-ID: 21cdaea43523056c3a09c45b13c9a940 at 189.6.70.47
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:3013 at 189.6.70.47>
Content-Length: 0


<------------>
[Nov 13 10:21:18] VERBOSE[25087] logger.c:
<--- Reliably Transmitting (NAT) to 189.160.126.210:5060 --->
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP
189.160.126.210:5060;branch=z9hG4bKb9f6b05aa5fb314af51ece37c;received=189.160.126.210
From: <sip:0411331644005 at 189.160.126.210>;tag=66359f37
To: "Teste"<sip:3013 at 189.6.70.47>;tag=as41b028c6
Call-ID: 21cdaea43523056c3a09c45b13c9a940 at 189.6.70.47
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16


<------------>
[Nov 13 10:21:18] VERBOSE[25087] logger.c:
<--- SIP read from 189.160.126.210:5060 --->
ACK sip:3013 at 189.6.70.47 SIP/2.0
Via: SIP/2.0/UDP 189.160.126.210:5060;branch=z9hG4bKb9f6b05aa5fb314af51ece37c
Call-ID: 21cdaea43523056c3a09c45b13c9a940 at 189.6.70.47
From: <sip:0411331644005 at 189.160.126.210>;tag=66359f37
To: "Teste"<sip:3013 at 189.6.70.47>;tag=as41b028c6
CSeq: 1 ACK
Max-Forwards: 70
Content-Length: 0

<------------->
[Nov 13 10:21:18] VERBOSE[25087] logger.c: --- (8 headers 0 lines) ---
[Nov 13 10:21:18] VERBOSE[25087] logger.c:
<--- SIP read from 189.160.126.210:5060 --->
BYE sip:3013 at 189.6.70.47 SIP/2.0
Via: SIP/2.0/UDP 189.160.126.210:5060;branch=z9hG4bK4cd5c1fb8ed03c364cf46fbbf
Call-ID: 21cdaea43523056c3a09c45b13c9a940 at 189.6.70.47
From: <sip:0411331644005 at 189.160.126.210>;tag=66359f37
To: "Teste"<sip:3013 at 189.6.70.47>;tag=as41b028c6
CSeq: 2 BYE
Max-Forwards: 70
Content-Length: 0


<------------->
[Nov 13 10:21:18] VERBOSE[25087] logger.c: --- (8 headers 0 lines) ---
[Nov 13 10:21:18] VERBOSE[25087] logger.c: Sending to 189.160.126.210 : 5060
(NAT)
[Nov 13 10:21:18] VERBOSE[25087] logger.c:
<--- Transmitting (NAT) to 189.160.126.210:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
189.160.126.210:5060;branch=z9hG4bK4cd5c1fb8ed03c364cf46fbbf;received=189.160.126.210
From: <sip:0411331644005 at 189.160.126.210>;tag=66359f37
To: "Teste"<sip:3013 at 189.6.70.47>;tag=as41b028c6
Call-ID: 21cdaea43523056c3a09c45b13c9a940 at 189.6.70.47
CSeq: 2 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0



thank in advance.

--
Marcus


     
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David Backeberg

2009-Nov-14 15:50 UTC

head link

[asterisk-users] Asterisk with T38 Fax

On Sat, Nov 14, 2009 at 10:20 AM, Marcus Vinicius
<marc_mcs10 at yahoo.com.br> wrote:> I'm trying to send faxes using Asterisk 1.4 and T38 with sip but
Asterisk rejects the t38.
>
> Anybody know if is possible to transmit t38 fax with Asterisk 1.4?
It might be possible to T.38 fax with 1.4, but I do not recommend it.

T.38 fax is much better in asterisk-1.6, and it is actively getting
improved with each release.

I recommend that you try upgrading to 1.6 if you want to do T.38 fax.

If you don't want to do a full upgrade, start with just the fax, and I
think you'll find that you like it better for other things as well.

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