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Сообщений: 15
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t38 + aster 1.6.1 - receive/send fax
Давно не смотрел в сторону T38 у астера.
И вот у нас повился VoIP провайдер с поддержкой T38.
На прод сервере стоит 1.4.17. Там не стал ковырять.
Решил попробовать свежее астер 1.6.1 и spandsp-0.0.6
Все собрал. И решил протестить с Zopier
Неудача.
Та же история с spandsp 0.0.5
Те же грабли с 1.6.0.9
В sip.con конечно включил t38pt_udptl = yes
ulaw c с обоих концов
Перерыл кучу инфи но так и не понял почему оно не работает.
Вот что имею.
Asterisk 1.6.0.9, Copyright (C) 1999 - 2008 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
Connected to Asterisk 1.6.0.9 currently running on tank (pid = 29195)
Verbosity is at least 100
tank*CLI>
tank*CLI>
tank*CLI>
<--- SIP read from UDP://10.10.0.188:5060 --->
INVITE sip:100@10.10.0.157;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.10.0.188:5060;branch=z9hG4bK-d8754z-b151a2e772484967-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:test@10.10.0.188:5060;transport=UDP>
To: <sip:100@10.10.0.157;transport=UDP>
From: <sip:test@10.10.0.157;transport=UDP>;tag=765ae221
Call-ID: ZTIxY2IwZDEyY2ZkMzEwMDQyZTFiMmY1ZTc2MDI4NGM.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO
Content-Type: application/sdp
User-Agent: Zoiper for Windows rev.2875
Content-Length: 201
v=0
o=Zoiper_user 0 0 IN IP4 10.10.0.188
s=Zoiper_user
c=IN IP4 10.10.0.188
t=0 0
m=audio 8000 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (12 headers 10 lines) ---
== Using SIP RTP CoS mark 5
== Using UDPTL CoS mark 5
Sending to 10.10.0.188 : 5060 (NAT)
Using INVITE request as basis request - ZTIxY2IwZDEyY2ZkMzEwMDQyZTFiMmY1ZTc2MDI4NGM.
Found user 'test' for 'test'
<--- Reliably Transmitting (no NAT) to 10.10.0.188:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.10.0.188:5060;branch=z9hG4bK-d8754z-b151a2e772484967-1---d8754z-;received=10.10.0.188;rport=5060
From: <sip:test@10.10.0.157;transport=UDP>;tag=765ae221
To: <sip:100@10.10.0.157;transport=UDP>;tag=as38d974a0
Call-ID: ZTIxY2IwZDEyY2ZkMzEwMDQyZTFiMmY1ZTc2MDI4NGM.
CSeq: 1 INVITE
User-Agent: Asterisk PBX 1.6.0.9
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2b2a44fc"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog 'ZTIxY2IwZDEyY2ZkMzEwMDQyZTFiMmY1ZTc2MDI4NGM.' in 32000 ms (Method: INVITE)
tank*CLI>
<--- SIP read from UDP://10.10.0.188:5060 --->
ACK sip:100@10.10.0.157;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.10.0.188:5060;branch=z9hG4bK-d8754z-b151a2e772484967-1---d8754z-;rport
To: <sip:100@10.10.0.157;transport=UDP>;tag=as38d974a0
From: <sip:test@10.10.0.157;transport=UDP>;tag=765ae221
Call-ID: ZTIxY2IwZDEyY2ZkMzEwMDQyZTFiMmY1ZTc2MDI4NGM.
CSeq: 1 ACK
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
tank*CLI>
<--- SIP read from UDP://10.10.0.188:5060 --->
INVITE sip:100@10.10.0.157;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.10.0.188:5060;branch=z9hG4bK-d8754z-8c0722886839b008-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:test@10.10.0.188:5060;transport=UDP>
To: <sip:100@10.10.0.157;transport=UDP>
From: <sip:test@10.10.0.157;transport=UDP>;tag=765ae221
Call-ID: ZTIxY2IwZDEyY2ZkMzEwMDQyZTFiMmY1ZTc2MDI4NGM.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO
Content-Type: application/sdp
User-Agent: Zoiper for Windows rev.2875
Authorization: Digest username="test",realm="asterisk",nonce="2b2a44fc",uri="sip:100@10.10.0.157;transport=UDP",response="977211f
ce0fadb5e8e974",algorithm=MD5
Content-Length: 201
v=0
o=Zoiper_user 0 0 IN IP4 10.10.0.188
s=Zoiper_user
c=IN IP4 10.10.0.188
t=0 0
m=audio 8000 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (13 headers 10 lines) ---
Sending to 10.10.0.188 : 5060 (NAT)
Using INVITE request as basis request - ZTIxY2IwZDEyY2ZkMzEwMDQyZTFiMmY1ZTc2MDI4NGM.
Found user 'test' for 'test'
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 10.10.0.188:8000
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 10.10.0.188:8000
Looking for 100 in default (domain 10.10.0.157)
list_route: hop: <sip:test@10.10.0.188:5060;transport=UDP>
<--- Transmitting (no NAT) to 10.10.0.188:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.10.0.188:5060;branch=z9hG4bK-d8754z-8c0722886839b008-1---d8754z-;received=10.10.0.188;rport=5060
From: <sip:test@10.10.0.157;transport=UDP>;tag=765ae221
To: <sip:100@10.10.0.157;transport=UDP>
Call-ID: ZTIxY2IwZDEyY2ZkMzEwMDQyZTFiMmY1ZTc2MDI4NGM.
CSeq: 2 INVITE
User-Agent: Asterisk PBX 1.6.0.9
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:100@10.10.0.157>
Content-Length: 0
<------------>
-- Executing [100@default:1] ReceiveFAX("SIP/test-09fc2f08", "/tmp/test.tiff") in new stack
Audio is at 10.10.0.157 port 18112
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Reliably Transmitting (no NAT) to 10.10.0.188:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.0.188:5060;branch=z9hG4bK-d8754z-8c0722886839b008-1---d8754z-;received=10.10.0.188;rport=5060
From: <sip:test@10.10.0.157;transport=UDP>;tag=765ae221
To: <sip:100@10.10.0.157;transport=UDP>;tag=as278bd328
Call-ID: ZTIxY2IwZDEyY2ZkMzEwMDQyZTFiMmY1ZTc2MDI4NGM.
CSeq: 2 INVITE
User-Agent: Asterisk PBX 1.6.0.9
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:100@10.10.0.157>
Content-Type: application/sdp
Content-Length: 261
v=0
o=root 1718651154 1718651154 IN IP4 10.10.0.157
s=Asterisk PBX 1.6.0.9
c=IN IP4 10.10.0.157
t=0 0
m=audio 18112 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------>
tank*CLI>
<--- SIP read from UDP://10.10.0.188:5060 --->
ACK sip:100@10.10.0.157 SIP/2.0
Via: SIP/2.0/UDP 10.10.0.188:5060;branch=z9hG4bK-d8754z-390d871a12b57aca-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:test@10.10.0.188:5060;transport=UDP>
To: <sip:100@10.10.0.157;transport=UDP>;tag=as278bd328
From: <sip:test@10.10.0.157;transport=UDP>;tag=765ae221
Call-ID: ZTIxY2IwZDEyY2ZkMzEwMDQyZTFiMmY1ZTc2MDI4NGM.
CSeq: 2 ACK
User-Agent: Zoiper for Windows rev.2875
Authorization: Digest username="test",realm="asterisk",nonce="2b2a44fc",uri="sip:100@10.10.0.157;transport=UDP",response="977211f
ce0fadb5e8e974",algorithm=MD5
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
tank*CLI>
<--- SIP read from UDP://10.10.0.188:5060 --->
INVITE sip:100@10.10.0.157 SIP/2.0
Via: SIP/2.0/UDP 10.10.0.188:5060;branch=z9hG4bK-d8754z-de6fcfb6d54d0475-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:test@10.10.0.188:5060;transport=UDP>
To: <sip:100@10.10.0.157;transport=UDP>;tag=as278bd328
From: <sip:test@10.10.0.157;transport=UDP>;tag=765ae221
Call-ID: ZTIxY2IwZDEyY2ZkMzEwMDQyZTFiMmY1ZTc2MDI4NGM.
CSeq: 3 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO
Content-Type: application/sdp
User-Agent: Zoiper for Windows rev.2875
Authorization: Digest username="test",realm="asterisk",nonce="2b2a44fc",uri="sip:100@10.10.0.157",response="b9afeb630dfb17e475162
",algorithm=MD5
Content-Length: 367
v=0
o=Zoiper_user 517578557 184362255 IN IP4 10.10.0.188
s=Zoiper_user
c=IN IP4 10.10.0.188
t=0 0
m=image 8000 udptl t38
a=T38FaxRateManagement:transferredTCF
a=T38FaxVersion:0
a=T38FaxMaxBuffer:400
a=T38FaxTranscodingMMR:0
a=T38FaxUdpEC:t38UDPRedundancy
a=T38FaxFillBitRemoval:0
a=T38MaxBitRate:14400
a=T38FaxMaxDatagram:400
a=T38FaxTranscodingJBIG:0
<------------->
--- (13 headers 15 lines) ---
Sending to 10.10.0.188 : 5060 (NAT)
Got T.38 offer in SDP in dialog ZTIxY2IwZDEyY2ZkMzEwMDQyZTFiMmY1ZTc2MDI4NGM.
Got T.38 Re-invite without audio. Keeping RTP active during T.38 session. Callid ZTIxY2IwZDEyY2ZkMzEwMDQyZTFiMmY1ZTc2MDI4NGM.
Capabilities: us - 0x4 (ulaw), 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)
<--- Transmitting (NAT) to 10.10.0.188:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.10.0.188:5060;branch=z9hG4bK-d8754z-de6fcfb6d54d0475-1---d8754z-;received=10.10.0.188;rport=5060
From: <sip:test@10.10.0.157;transport=UDP>;tag=765ae221
To: <sip:100@10.10.0.157;transport=UDP>;tag=as278bd328
Call-ID: ZTIxY2IwZDEyY2ZkMzEwMDQyZTFiMmY1ZTc2MDI4NGM.
CSeq: 3 INVITE
User-Agent: Asterisk PBX 1.6.0.9
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:100@10.10.0.157>
Content-Length: 0
tank*CLI>
<--- Reliably Transmitting (NAT) to 10.10.0.188:5060 --->
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP 10.10.0.188:5060;branch=z9hG4bK-d8754z-de6fcfb6d54d0475-1---d8754z-;received=10.10.0.188;rport=5060
From: <sip:test@10.10.0.157;transport=UDP>;tag=765ae221
To: <sip:100@10.10.0.157;transport=UDP>;tag=as278bd328
Call-ID: ZTIxY2IwZDEyY2ZkMzEwMDQyZTFiMmY1ZTc2MDI4NGM.
CSeq: 3 INVITE
User-Agent: Asterisk PBX 1.6.0.9
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Length: 0
<------------>
[May 7 22:56:56] ERROR[29365]: channel.c:2520 __ast_read: ast_read() called with no recorded file descriptor.
[May 7 22:56:56] ERROR[29365]: channel.c:2520 __ast_read: ast_read() called with no recorded file descriptor.
[May 7 22:56:56] ERROR[29365]: channel.c:2520 __ast_read: ast_read() called with no recorded file descriptor.
tank*CLI>
<--- SIP read from UDP://10.10.0.188:5060 --->
ACK sip:100@10.10.0.157 SIP/2.0
Via: SIP/2.0/UDP 10.10.0.188:5060;branch=z9hG4bK-d8754z-de6fcfb6d54d0475-1---d8754z-;rport
To: <sip:100@10.10.0.157;transport=UDP>;tag=as278bd328
From: <sip:test@10.10.0.157;transport=UDP>;tag=765ae221
Call-ID: ZTIxY2IwZDEyY2ZkMzEwMDQyZTFiMmY1ZTc2MDI4NGM.
CSeq: 3 ACK
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
tank*CLI>
<--- SIP read from UDP://10.10.0.188:5060 --->
BYE sip:100@10.10.0.157 SIP/2.0
Via: SIP/2.0/UDP 10.10.0.188:5060;branch=z9hG4bK-d8754z-f3bc204a253cbd65-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:test@10.10.0.188:5060;transport=UDP>
To: <sip:100@10.10.0.157;transport=UDP>;tag=as278bd328
From: <sip:test@10.10.0.157;transport=UDP>;tag=765ae221
Call-ID: ZTIxY2IwZDEyY2ZkMzEwMDQyZTFiMmY1ZTc2MDI4NGM.
CSeq: 4 BYE
User-Agent: Zoiper for Windows rev.2875
Authorization: Digest username="test",realm="asterisk",nonce="2b2a44fc",uri="sip:100@10.10.0.157",response="a29fcc93c2b2e9fda666a
",algorithm=MD5
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
Sending to 10.10.0.188 : 5060 (NAT)
<--- Transmitting (NAT) to 10.10.0.188:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.0.188:5060;branch=z9hG4bK-d8754z-f3bc204a253cbd65-1---d8754z-;received=10.10.0.188;rport=5060
From: <sip:test@10.10.0.157;transport=UDP>;tag=765ae221
To: <sip:100@10.10.0.157;transport=UDP>;tag=as278bd328
Call-ID: ZTIxY2IwZDEyY2ZkMzEwMDQyZTFiMmY1ZTc2MDI4NGM.
CSeq: 4 BYE
User-Agent: Asterisk PBX 1.6.0.9
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Length: 0
<------------>
[May 7 22:56:56] WARNING[29365]: app_fax.c:178 phase_e_handler: Error transmitting fax. result=49: The call dropped prematurely.
[May 7 22:56:56] WARNING[29365]: app_fax.c:650 transmit: Transmission error
== Spawn extension (default, 100, 1) exited non-zero on 'SIP/test-09fc2f08'
Really destroying SIP dialog 'ZTIxY2IwZDEyY2ZkMzEwMDQyZTFiMmY1ZTc2MDI4NGM.' Method: BYE
tank*CLI>
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