我尝试在正在运行的 Debian 10 服务器上安装 HylaFax + IAXModem + Asterisk。服务器直接连接到 Internet，防火墙已禁用（未设置规则）。服务器唯一要做的就是发送传真。我使用 sipgate 中继。在 HylaFax 尝试发送传真期间，我在 Asterisk 控制台中看到此输出（SIP 调试和详细程度已打开/10）：
-- Accepting AUTHENTICATED call from 127.0.0.1:4570:
-- > requested format = alaw,
-- > requested prefs = (),
-- > actual format = alaw,
-- > host prefs = (alaw),
-- > priority = mine
-- Executing [RECIPIENT@fax_out:1] Set("IAX2/iaxmodem-7708", "CALLERID(num)=CALLER") in new stack
-- Executing [RECIPIENT@fax_out:2] Set("IAX2/iaxmodem-7708", "CALLERID(name)=CALLER") in new stack
-- Executing [RECIPIENT@fax_out:3] SIPAddHeader("IAX2/iaxmodem-7708", "P-Preferred-Identity:<sip:CALLER>") in new stack
-- Executing [RECIPIENT@fax_out:4] Dial("IAX2/iaxmodem-7708", "SIP/sipconnect.sipgate.de/RECIPIENT,,r") in new stack
== Using SIP RTP CoS mark 5
Audio is at 17702
Adding codec alaw to SDP
Adding codec ulaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 217.10.68.150:5060:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 127.0.1.1:5060;branch=z9hG4bK12349644
Max-Forwards: 70
From: "CALLER" <sip:[email protected]>;tag=as65bc91b0
To: <sip:[email protected]:5060>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX 16.2.1~dfsg-1+deb10u1
Date: Mon, 21 Oct 2019 12:22:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
P-Preferred-Identity: <sip:CALLER>
P-Asserted-Identity: "CALLER" <sip:[email protected]>
Content-Type: application/sdp
Content-Length: 270
v=0
o=root 698738770 698738770 IN IP4 127.0.1.1
s=Asterisk PBX 16.2.1~dfsg-1+deb10u1
c=IN IP4 127.0.1.1
t=0 0
m=audio 17702 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv
---
-- Called SIP/sipconnect.sipgate.de/RECIPIENT
<--- SIP read from UDP:217.10.68.150:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 127.0.1.1:5060;rport=5060;received=X.X.X.X;branch=z9hG4bK12349644
From: "CALLER" <sip:[email protected]>;tag=as65bc91b0
To: <sip:[email protected]:5060>;tag=c63713a666d5644779294882ed89253a.0c69
Call-ID: [email protected]
CSeq: 102 INVITE
Proxy-Authenticate: Digest realm="sipconnect.sipgate.de", nonce="Xa2kKF2tovxRNF/AcCiPaUlB/z/ev7jl"
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
Transmitting (no NAT) to 217.10.68.150:5060:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 127.0.1.1:5060;branch=z9hG4bK12349644
Max-Forwards: 70
From: "CALLER" <sip:[email protected]>;tag=as65bc91b0
To: <sip:[email protected]:5060>;tag=c63713a666d5644779294882ed89253a.0c69
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 102 ACK
User-Agent: Asterisk PBX 16.2.1~dfsg-1+deb10u1
Content-Length: 0
---
Audio is at 17702
Adding codec alaw to SDP
Adding codec ulaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 217.10.68.150:5060:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 127.0.1.1:5060;branch=z9hG4bK248f1ffc
Max-Forwards: 70
From: "CALLER" <sip:[email protected]>;tag=as65bc91b0
To: <sip:[email protected]:5060>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 103 INVITE
User-Agent: Asterisk PBX 16.2.1~dfsg-1+deb10u1
Proxy-Authorization: Digest username="USER", realm="sipconnect.sipgate.de", algorithm=MD5, uri="sip:[email protected]:5060", nonce="Xa2kKF2tovxRNF/AcCiPaUlB/z/ev7jl", response="42016e991f588a252062bb86b35a3f6c"
Date: Mon, 21 Oct 2019 12:22:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
P-Preferred-Identity: <sip:CALLER>
P-Asserted-Identity: "CALLER" <sip:[email protected]>
Content-Type: application/sdp
Content-Length: 270
v=0
o=root 698738770 698738771 IN IP4 127.0.1.1
s=Asterisk PBX 16.2.1~dfsg-1+deb10u1
c=IN IP4 127.0.1.1
t=0 0
m=audio 17702 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv
---
<--- SIP read from UDP:217.10.68.150:5060 --->
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 127.0.1.1:5060;rport=5060;received=X.X.X.X;branch=z9hG4bK248f1ffc
From: "CALLER" <sip:[email protected]>;tag=as65bc91b0
To: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 103 INVITE
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
<--- SIP read from UDP:217.10.68.150:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 127.0.1.1:5060;rport=5060;received=X.X.X.X;branch=z9hG4bK248f1ffc
Record-Route: <sip:172.20.40.8;lr>
Record-Route: <sip:217.10.68.150;lr;ftag=as65bc91b0>
From: "CALLER" <sip:[email protected]>;tag=as65bc91b0
To: <sip:[email protected]:5060>;tag=as21618100
Call-ID: [email protected]
CSeq: 103 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Contact: <sip:[email protected]:5060>
Content-Length: 0
<------------->
--- (12 headers 0 lines) ---
sip_route_dump: route/path hop: <sip:217.10.68.150;lr;ftag=as65bc91b0>
sip_route_dump: route/path hop: <sip:172.20.40.8;lr>
-- SIP/sipconnect.sipgate.de-00000003 is ringing
<--- SIP read from UDP:217.10.68.150:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 127.0.1.1:5060;rport=5060;received=X.X.X.X;branch=z9hG4bK248f1ffc
Record-Route: <sip:172.20.40.8;lr>
Record-Route: <sip:217.10.68.150;lr;ftag=as65bc91b0>
From: "CALLER" <sip:[email protected]>;tag=as65bc91b0
To: <sip:[email protected]:5060>;tag=as21618100
Call-ID: [email protected]
CSeq: 103 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Contact: <sip:[email protected]:5060>
Content-Type: application/sdp
Content-Length: 298
v=0
o=root 1290589385 1290589385 IN IP4 217.116.117.70
s=sipgate VoIP GW
c=IN IP4 212.9.44.253
t=0 0
m=audio 15550 RTP/AVP 8 0 101
a=silenceSupp:off - - - -
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
a=rtcp:15551
a=ptime:20
<------------->
--- (13 headers 14 lines) ---
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - (alaw|ulaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (alaw|ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
> 0x7f4634015640 -- Strict RTP learning after remote address set to: 212.9.44.253:15550
Peer audio RTP is at port 212.9.44.253:15550
sip_route_dump: route/path hop: <sip:217.10.68.150;lr;ftag=as65bc91b0>
sip_route_dump: route/path hop: <sip:172.20.40.8;lr>
set_destination: Parsing <sip:217.10.68.150;lr;ftag=as65bc91b0> for address/port to send to
set_destination: set destination to 217.10.68.150:5060
Transmitting (no NAT) to 217.10.68.150:5060:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 127.0.1.1:5060;branch=z9hG4bK12463de5
Route: <sip:217.10.68.150;lr;ftag=as65bc91b0>,<sip:172.20.40.8;lr>
Max-Forwards: 70
From: "CALLER" <sip:[email protected]>;tag=as65bc91b0
To: <sip:[email protected]:5060>;tag=as21618100
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 103 ACK
User-Agent: Asterisk PBX 16.2.1~dfsg-1+deb10u1
Content-Length: 0
---
-- SIP/sipconnect.sipgate.de-00000003 answered IAX2/iaxmodem-7708
-- Channel SIP/sipconnect.sipgate.de-00000003 joined 'simple_bridge' basic-bridge <262f871a-8fc7-4bb9-a64b-981378a84acd>
-- Channel IAX2/iaxmodem-7708 joined 'simple_bridge' basic-bridge <262f871a-8fc7-4bb9-a64b-981378a84acd>
> 0x7f4634015640 -- Strict RTP switching to RTP target address 212.9.44.253:15550 as source
<--- SIP read from UDP:217.10.68.150:5060 --->
<------------->
<--- SIP read from UDP:217.10.68.150:5060 --->
BYE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 217.10.68.150;branch=z9hG4bKd711.674d11475dd9179e68c4eb52c2088642.0
Via: SIP/2.0/UDP 172.20.40.8;branch=z9hG4bKd711.39ae43c7fc3d82443eba26f3d75b5f39.0
Via: SIP/2.0/UDP 217.116.117.70:5060;branch=z9hG4bK5d2cbd05
Max-Forwards: 68
From: <sip:[email protected]:5060>;tag=as21618100
To: "CALLER" <sip:[email protected]>;tag=as65bc91b0
Call-ID: [email protected]
CSeq: 102 BYE
Reason: Q.850;cause=16
Content-Length: 0
X-hint: rr-enforced
<------------->
--- (12 headers 0 lines) ---
Sending to 217.10.68.150:5060 (no NAT)
Scheduling destruction of SIP dialog '[email protected]' in 6400 ms (Method: BYE)
<--- Transmitting (no NAT) to 217.10.68.150:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 217.10.68.150;branch=z9hG4bKd711.674d11475dd9179e68c4eb52c2088642.0;received=217.10.68.150
Via: SIP/2.0/UDP 172.20.40.8;branch=z9hG4bKd711.39ae43c7fc3d82443eba26f3d75b5f39.0
Via: SIP/2.0/UDP 217.116.117.70:5060;branch=z9hG4bK5d2cbd05
From: <sip:[email protected]:5060>;tag=as21618100
To: "CALLER" <sip:[email protected]>;tag=as65bc91b0
Call-ID: [email protected]
CSeq: 102 BYE
Server: Asterisk PBX 16.2.1~dfsg-1+deb10u1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
<------------>
-- Channel SIP/sipconnect.sipgate.de-00000003 left 'simple_bridge' basic-bridge <262f871a-8fc7-4bb9-a64b-981378a84acd>
-- Channel IAX2/iaxmodem-7708 left 'simple_bridge' basic-bridge <262f871a-8fc7-4bb9-a64b-981378a84acd>
== Spawn extension (fax_out, RECIPIENT, 4) exited non-zero on 'IAX2/iaxmodem-7708'
-- Hungup 'IAX2/iaxmodem-7708'
Really destroying SIP dialog '[email protected]' Method: BYE
在/var/spool/hylafax/log/xferfaxlog：
10/21/19 15:38 SEND 000000096 ttyIAX0 41 "" [email protected] "RECIPIENT" "" 2220072 0 0:00:03 0:00:00 "No carrier detected" "" "" "" "root" "00 00 00"
（我在这里替换了发件人/收件人号码和用户名）
防火墙当前未运行：
root@asterisk:/etc/asterisk# iptables -nL
Chain INPUT (policy ACCEPT)
target prot opt source destination
Chain FORWARD (policy ACCEPT)
target prot opt source destination
Chain OUTPUT (policy ACCEPT)
target prot opt source destination
这是我的 Asterisk 配置：
在/etc/asterisk/sip.conf：
[general]
context=unauthenticated
bindport=5060
bindaddr=0.0.0.0
realm=domain.tld
externhost=domain.tld:5060
localnet=127.0.0.0/255.255.255.0
nat=no
srvlookup=yes
allowguest=no
alwaysauthreject=yes
register => USER:[email protected]/USER
[sipconnect.sipgate.de]
type=peer
host=sipconnect.sipgate.de
outboundproxy=sipconnect.sipgate.de
port=5060
username=USER
defaultuser=USER
fromuser=USER
fromdomain=sipconnect.sipgate.de
secret=PASS
dtmfmode=rfc2833
insecure=port,invite
canreinvite=no
directmedia=no
registertimeout=600
sendrpid=pai
usereqphone=no
t38pt_udptl=no
disallow=all
allow=alaw
allow=ulaw
qualify=yes
context=unauthenticated
（我在这里替换了凭证和域名）
在/etc/asterisk/extensions.conf：
[general]
[sipgate_in]
exten => sipgate_in,1,Goto(siptrunk,${CUT(CUT(SIP_HEADER(To),@,1),:,2)},1)
[siptrunk]
exten => 1234567,1,Dial(IAX2/iaxmodem)
exten => 1234567,n,Hangup
[fax_out]
exten => _X.,1,Set(CALLERID(num)=00491231234567)
exten => _X.,n,Set(CALLERID(name)=${CALLERID(num)})
exten => _X.,n,SipAddHeader(P-Preferred-Identity:<sip:${CALLERID(num)}>)
exten => _X.,n,Dial(SIP/sipconnect.sipgate.de/${EXTEN},,r)
[unauthenticated]
（我在这里替换了发件人号码）
在/etc/asterisk/iax.conf：
[general]
bindport=4569
bindaddr=127.0.0.1
calltokenoptional=127.0.0.1/255.255.255.0
[iaxmodem]
port=4570
type=friend
host=dynamic
qualify=yes
secret=pwd
requirecalltoken=no
disallow=all
allow=alaw
jitterbuffer=no
trunk=no
context=fax_out
（我在这里替换了凭证）
在/etc/iaxmodem/ttyIAX0：
device /dev/ttyIAX0
owner uucp:uucp
mode 660
port 4570
refresh 60
server 127.0.0.1
peername iaxmodem
secret pwd
codec alaw
nojitterbuffer
（我在这里替换了凭证）
IAXModemttyIAX0确实成功注册，Asterisk 在 sipgate 中继上处于在线状态。从已知工作设置向收件人发送传真成功。之前我收到了一些网络协议错误，但由于我在测试期间没有激活防火墙，这些错误来自尝试注册为设备等的某些地方...
接收方正在运行不使用 T.38 的 3CX，因此我也在配置中禁用了 T.38，只是为了确保 T.38 不是问题。
据我了解，调试输出显示目标设备在发送传真之前挂断。我理解的对吗？有人知道为什么通信会这样吗？我如何才能找到提前挂断的原因？
更新：我现在能够向另一个目标号码发送传真。也许我做对了所有事情，但 3CX 传真出现了问题。但我仍然不确定调试协议 - 在这种情况下它看起来应该如此吗？
更新 2：目标 3CX 已启用 T.38 回退，现在它从我的 Asterisk PBX 接收传真。我不太了解 3CX 配置，也不知道“回退”是什么意思 - 但是，现在它可以正常工作了。我仍然很好奇如何才能找到早期挂断的原因 - 也许根本就不可能。我希望我的配置现在对现实生活有用...
答案1
现在通过 SIP 调试，我发现您有单向音频问题（您这边的 RTP 发送您的本地主机 IP）。尝试启用nat=yes此中继sipconnect.sipgate.de。在 SIP 调试中，您应该看到您的公共 IP 被发送到 RDP：c=IN IP4 127.0.1.1应该是：c=IN IP4 your.public.ip.address。
对于受到攻击的 IP，我建议不要更改默认端口，而是启用防火墙并仅允许从sipconnect.sipgate.de（通常，提供商会为您提供 IP 列表或范围）到您的 SIP 端口的流量。