Centos PBX似乎正在拒绝（603）所有SIP呼叫

对于一个简单的家庭通信系统，我设置了一些非常简单的SIP/扩展。对我放松点，我对这个系统很陌生

目前，我让它们工作的唯一方法（在测试中）是拆除防火墙。尽管如此，每一部手机的每一次尝试似乎都让我获得了即时的603

当我打电话时，它会这样报告：

<--- SIP read from UDP:192.168.1.8:5060 --->
INVITE sip:103@192.168.1.6 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.8:5060;rport;branch=z9hG4bKPj43e20551-6dbc-411b-919a-ab117c06ae05
Max-Forwards: 70
From: <sip:0000FFFF004@192.168.1.6>;tag=12e33b02-c7c1-4b83-9ba4-f08c426ece25
To: <sip:103@192.168.1.6>
Contact: <sip:0000FFFF004@192.168.1.8:5060>
Call-ID: 16962f1e-d2e0-4987-af02-dc765ffa793f
CSeq: 6702 INVITE
Allow: PRACK, SUBSCRIBE, NOTIFY, REFER, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, REGISTER, OPTIONS, MESSAGE
upported: replaces, 100rel
Content-Type: application/sdp
Content-Length: 361

v=0
o=dinosaur 3611940779 0 IN IP4 192.168.1.8
s=sflphone
c=IN IP4 192.168.1.8
t=0 0
m=audio 37600 RTP/AVP 0 3 8 9 110 111 112
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:110 speex/8000
a=rtpmap:111 speex/16000
a=rtpmap:112 speex/32000
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
--- (12 headers 16 lines) ---
Sending to 192.168.1.8:5060 (NAT)
Using INVITE request as basis request - 16962f1e-d2e0-4987-af02-dc765ffa793f
Found peer '0000FFFF004' for '0000FFFF004' from 192.168.1.8:5060

<--- Reliably Transmitting (NAT) to 192.168.1.8:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.8:5060;branch=z9hG4bKPj43e20551-6dbc-411b-919a-ab117c06ae05;received=192.168.1.8;rport=5060
From: <sip:0000FFFF004@192.168.1.6>;tag=12e33b02-c7c1-4b83-9ba4-f08c426ece25
To: <sip:103@192.168.1.6>;tag=as69cdb064
Call-ID: 16962f1e-d2e0-4987-af02-dc765ffa793f
CSeq: 6702 INVITE
Server: Asterisk PBX SVN-branch-1.8-r416150
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5572b5df"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '16962f1e-d2e0-4987-af02-dc765ffa793f' in 32000 ms (Method: INVITE)

<--- SIP read from UDP:192.168.1.8:5060 --->
ACK sip:103@192.168.1.6 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.8:5060;rport;branch=z9hG4bKPj43e20551-6dbc-411b-919a-ab117c06ae05
Max-Forwards: 70
From: <sip:0000FFFF004@192.168.1.6>;tag=12e33b02-c7c1-4b83-9ba4-f08c426ece25
To: <sip:103@192.168.1.6>;tag=as69cdb064
Call-ID: 16962f1e-d2e0-4987-af02-dc765ffa793f
CSeq: 6702 ACK
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---

<--- SIP read from UDP:192.168.1.8:5060 --->
INVITE sip:103@192.168.1.6 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.8:5060;rport;branch=z9hG4bKPj7504ac55-fcbe-470a-a5c5-2174a0699d0d
Max-Forwards: 70
From: <sip:0000FFFF004@192.168.1.6>;tag=12e33b02-c7c1-4b83-9ba4-f08c426ece25
To: <sip:103@192.168.1.6>
Contact: <sip:0000FFFF004@192.168.1.8:5060>
Call-ID: 16962f1e-d2e0-4987-af02-dc765ffa793f
CSeq: 6703 INVITE
Allow: PRACK, SUBSCRIBE, NOTIFY, REFER, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, REGISTER, OPTIONS, MESSAGE
upported: replaces, 100rel
Authorization: Digest username="0000FFFF004", realm="asterisk", nonce="5572b5df", uri="sip:103@192.168.1.6", response="44810c7fbf0d8a99e34ea07b5e62ee79", algorithm=MD5
Content-Type: application/sdp
Content-Length: 361

v=0
o=dinosaur 3611940779 0 IN IP4 192.168.1.8
s=sflphone
c=IN IP4 192.168.1.8
t=0 0
m=audio 37600 RTP/AVP 0 3 8 9 110 111 112
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:110 speex/8000
a=rtpmap:111 speex/16000
a=rtpmap:112 speex/32000
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
--- (13 headers 16 lines) ---
Sending to 192.168.1.8:5060 (NAT)
Using INVITE request as basis request - 16962f1e-d2e0-4987-af02-dc765ffa793f
Found peer '0000FFFF004' for '0000FFFF004' from 192.168.1.8:5060
  == Using SIP RTP CoS mark 5
Found RTP audio format 0
Found RTP audio format 3
Found RTP audio format 8
Found RTP audio format 9
Found RTP audio format 110
Found RTP audio format 111
Found RTP audio format 112
Found audio description format PCMU for ID 0
Found audio description format GSM for ID 3
Found audio description format PCMA for ID 8
Found audio description format G722 for ID 9
Found audio description format speex for ID 110
Found audio description format speex for ID 111
Found unknown media description format speex for ID 112
Found audio description format telephone-event for ID 101
Capabilities: us - 0x100e (gsm|ulaw|alaw|g722), peer - audio=0x20000120e (gsm|ulaw|alaw|speex|speex16|g722)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x100e (gsm|ulaw|alaw|g722)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.1.8:37600
Looking for 103 in LocalSets (domain 192.168.1.6)
list_route: hop: <sip:0000FFFF004@192.168.1.8:5060>

<--- Transmitting (NAT) to 192.168.1.8:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.8:5060;branch=z9hG4bKPj7504ac55-fcbe-470a-a5c5-2174a0699d0d;received=192.168.1.8;rport=5060
From: <sip:0000FFFF004@192.168.1.6>;tag=12e33b02-c7c1-4b83-9ba4-f08c426ece25
To: <sip:103@192.168.1.6>
Call-ID: 16962f1e-d2e0-4987-af02-dc765ffa793f
CSeq: 6703 INVITE
Server: Asterisk PBX SVN-branch-1.8-r416150
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:103@192.168.1.6:5060>
Content-Length: 0


<------------>
    -- Executing [103@LocalSets:1] Dial("SIP/0000FFFF004-0000001a", "0000FFFF005") in new stack
  == Spawn extension (LocalSets, 103, 1) exited non-zero on 'SIP/0000FFFF004-0000001a'
Scheduling destruction of SIP dialog '16962f1e-d2e0-4987-af02-dc765ffa793f' in 32000 ms (Method: INVITE)

<--- Reliably Transmitting (NAT) to 192.168.1.8:5060 --->
SIP/2.0 603 Declined
Via: SIP/2.0/UDP 192.168.1.8:5060;branch=z9hG4bKPj7504ac55-fcbe-470a-a5c5-2174a0699d0d;received=192.168.1.8;rport=5060
From: <sip:0000FFFF004@192.168.1.6>;tag=12e33b02-c7c1-4b83-9ba4-f08c426ece25
To: <sip:103@192.168.1.6>;tag=as165ecdc9
Call-ID: 16962f1e-d2e0-4987-af02-dc765ffa793f
CSeq: 6703 INVITE
Server: Asterisk PBX SVN-branch-1.8-r416150
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------>

<--- SIP read from UDP:192.168.1.8:5060 --->
ACK sip:103@192.168.1.6 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.8:5060;rport;branch=z9hG4bKPj7504ac55-fcbe-470a-a5c5-2174a0699d0d
Max-Forwards: 70
From: <sip:0000FFFF004@192.168.1.6>;tag=12e33b02-c7c1-4b83-9ba4-f08c426ece25
To: <sip:103@192.168.1.6>;tag=as165ecdc9
Call-ID: 16962f1e-d2e0-4987-af02-dc765ffa793f
CSeq: 6703 ACK
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---

<--- SIP read from UDP:192.168.1.5:63992 --->

<------------->
Really destroying SIP dialog 'cb7123d1-4244-4673-a200-dc851e1c8415' Method: REGISTER

邀请sip：103@192.168.1.6SIP/2.0
Via:SIP/2.0/UDP 192.168.1.8:5060；rport；分支机构=z9hG4bKPj43e20551-6dbc-411b-919a-ab117c06ae05
最大前锋：70
发件人：；标签=12e33b02-c7c1-4b83-9ba4-F08C42625
致：
联系人：
电话号码：16962f1e-d2e0-4987-af02-dc765ffa793f
CSeq:6702邀请
允许：PRACK、SUBSCRIBE、NOTIFY、REFER、INVITE、ACK、BYE、CANCEL、UPDATE、INFO、REGISTER、OPTIONS、MESSAGE
支持：替换，100rel
内容类型：应用程序/sdp
内容长度：361
v=0
o=恐龙3611940779 0，IP4 192.168.1.8
s=sflphone
c=在IP4 192.168.1.8中
t=0
m=音频37600 RTP/AVP 0 3 8 9 110 111 112
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:110 speex/8000
a=rtpmap:111 speex/16000
a=rtpmap:112 speex/32000
a=sendrecv
a=rtpmap:101电话事件/8000
a=fmtp:101 0-15
---（12页眉16行）---
发送至192.168.1.8:5060（NAT）
使用邀请请求作为基础请求-16962f1e-d2e0-4987-af02-dc765ffa793f
从192.168.1.8:5060找到“0000FFFF004”的对等方“0000FFFF004”
SIP/2.0 401未经授权
Via:SIP/2.0/UDP 192.168.1.8:5060；分支机构=z9hG4bKPj43e20551-6dbc-411b-919a-ab117c06ae05；接收=192.168.1.8；rport=5060
发件人：；标签=12e33b02-c7c1-4b83-9ba4-F08C42625
致：；标签=as69cdb064
电话号码：16962f1e-d2e0-4987-af02-dc765ffa793f
CSeq:6702邀请
服务器：星号PBX SVN-branch-1.8-r416150
允许：邀请、确认、取消、选项、再见、参考、订阅、通知、信息、发布、消息
支持：替换、定时器
WWW-Authenticate:Digest algorithm=MD5，realm=“asterisk”，nonce=“5572b5df”
内容长度：0
计划在32000毫秒内销毁SIP对话框“16962f1e-d2e0-4987-af02-dc765ffa793f”（方法：INVITE）
确认sip：103@192.168.1.6SIP/2.0
Via:SIP/2.0/UDP 192.168.1.8:5060；rport；分支机构=z9hG4bKPj43e20551-6dbc-411b-919a-ab117c06ae05
最大前锋：70
发件人：；标签=12e33b02-c7c1-4b83-9ba4-F08C42625
致：；标签=as69cdb064
电话号码：16962f1e-d2e0-4987-af02-dc765ffa793f
CSeq:6702确认
内容长度：0
---（8个标题0行）---
邀请sip：103@192.168.1.6SIP/2.0
Via:SIP/2.0/UDP 192.168.1.8:5060；rport；支管=z9hG4bKPj7504ac55-fcbe-470a-a5c5-2174a0699d0d
最大前锋：70
发件人：；标签=12e33b02-c7c1-4b83-9ba4-F08C42625
致：
联系人：
电话号码：16962f1e-d2e0-4987-af02-dc765ffa793f
CSeq:6703邀请
允许：PRACK、SUBSCRIBE、NOTIFY、REFER、INVITE、ACK、BYE、CANCEL、UPDATE、INFO、REGISTER、OPTIONS、MESSAGE
支持：替换，100rel
授权：摘要username=“0000FFFF004”，realm=“asterisk”，nonce=“5572b5df”，uri=“sip:103@192.168.1.6，response=“44810c7fbf0d8a99e34ea07b5e62ee79”，算法=MD5
内容类型：应用程序/sdp
内容长度：361
v=0
o=恐龙3611940779 0，IP4 192.168.1.8
s=sflphone
c=在IP4 192.168.1.8中
t=0
m=音频37600 RTP/AVP 0 3 8 9 110 111 112
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:110 speex/8000
a=rtpmap:111 speex/16000
a=rtpmap:112 speex/32000
a=sendrecv
a=rtpmap:101电话事件/8000
a=fmtp:101 0-15
---（13页眉16行）---
发送至192.168.1.8:5060（NAT）
使用邀请请求作为基础请求-16962f1e-d2e0-4987-af02-dc765ffa793f
从192.168.1.8:5060找到“0000FFFF004”的对等方“0000FFFF004”
==使用SIP RTP CoS标记5
找到RTP音频格式0
找到RTP音频格式3
找到RTP音频格式8
找到RTP音频格式9
找到RTP音频格式110
找到RTP音频格式111
找到RTP音频格式112
找到ID为0的音频描述格式PCMU
找到ID为3的GSM音频描述格式
找到ID为8的音频描述格式PCMA
找到ID 9的音频描述格式G722
找到ID为110的音频描述格式speex
找到ID为111的音频描述格式speex
找到ID为112的未知媒体描述格式speex
找到ID 101的音频描述格式电话事件
功能：us-0x100e（gsm | ulaw | alaw | g722），对等音频=0x20000120e（gsm | ulaw | alaw | speex | speex16 | g722）/视频=0x0（无）/文本=0x0（无），组合-0x100e（gsm | ulaw | alaw | g722）
非编解码器功能（dtmf）：us-0x1（电话事件|）、对等-0x1（电话事件|）、组合-0x1（电话事件|）
对等音频RTP位于端口192.168.1.8:37600
在localset（域192.168.1.6）中查找103
列表\u路由：跃点：
SIP/2.0 100
Via:SIP/2.0/UDP 192.168.1.8:5060；分支机构=z9hG4bKPj7504ac55-fcbe-470a-a5c5-2174a0699d0d；接收=192.168.1.8；rport=5060
发件人：；标签=12e33b02-c7c1-4b83-9ba4-F08C42625
致：
电话号码：16962f1e-d2e0-4987-af02-dc765ffa793f
CSeq:6703邀请
服务器：星号PBX SVN-branch-1.8-r416150
允许：邀请、确认、取消、选项、再见、参考、订阅、通知、信息、发布、消息
支持：替换、定时器
联系人：
内容长度：0
--执行[103@LocalSets：1]在新堆栈中拨号（“SIP/0000FFFF004-0000001a”，“0000FFFF005”）
==在“SIP/0000FFFF004-0000001a”上，生成扩展（localset，103，1）退出非零
计划在32000毫秒内销毁SIP对话框“16962f1e-d2e0-4987-af02-dc765ffa793f”（方法：INVITE）
SIP/2.0 603被拒绝
Via:SIP/2.0/UDP 192.168.1.8:5060；分支机构=z9hG4bKPj7504ac55-fcbe-470a-a5c5-2174a0699d0d；接收=192.168.1.8；rport=5060
发件人：；标签=12e33b02-c7c1-4b83-9ba4-F08C42625
致：；标签=as165ecdc9
电话号码：16962f1e-d2e0-4987-af02-dc765ffa793f
CSeq:6703邀请
服务器：星号PBX SVN-branch-1.8-r416150
允许：邀请、确认、取消、选项、再见、参考、订阅、通知、信息、发布、消息
支持：替换、定时器
内容长度：0
确认sip：103@192.168.1.6SIP/2.0
Via:SIP/2.0/UDP 192.168.1.8:5060；rport；支管=z9hG4bKPj7504ac55-fcbe-470a-a5c5-2174a0699d0d
最大前锋：70
发件人：；标签=12e33b02-c7c1-4b83-9ba4-F08C42625
致：；标签=as165ecdc9
电话号码：16962f