这是我第一次使用星号(基本上我什么都不知道，所以请原谅我)
我正在运行Asterisk 11.6在一个虚拟机512/kbps的互联网连接，这是背后的NAT。
有两个分机1001和1002，这是发生在我身上的情况。
号码1:使用软电话在本地呼叫。"没问题。"
号码2:从外部(软电话)呼叫本地工厂。"没问题。"
3号:从本地打到外面，很快就挂了。"问题"
。第4条:从外部呼叫到外部，从来没有成功过。我能听到拨号音，但接收器没有回应。"问题"
。我尝试转发端口5060 tcp和udp没有任何变化…
我也在某处读到我有NAT环回错误，在这一点上它不关心我。
我的问题是我想从外部网络连接这两个扩展…
(1001) Network1——>(服务器)Network2——> (1002)Network3
同样向后……我错过什么了吗?
这是我的sip配置。
Global Settings:
----------------
UDP Bindaddress: 0.0.0.0:5060
TCP SIP Bindaddress: Disabled
TLS SIP Bindaddress: Disabled
Videosupport: No
Textsupport: No
Ignore SDP sess. ver.: No
AutoCreate Peer: Off
Match Auth Username: No
Allow unknown access: Yes
Allow subscriptions: Yes
Allow overlap dialing: Yes
Allow promisc. redir: No
Enable call counters: No
SIP domain support: No
Realm. auth: No
Our auth realm asterisk
Use domains as realms: No
Call to non-local dom.: Yes
URI user is phone no: No
Always auth rejects: Yes
Direct RTP setup: No
User Agent: FPBX-AsteriskNOW-12.0.76(11.16.0)
SDP Session Name: Asterisk PBX 11.16.0
SDP Owner Name: root
Reg. context: (not set)
Regexten on Qualify: No
Trust RPID: No
Send RPID: No
Legacy userfield parse: No
Send Diversion: Yes
Caller ID: Unknown
From: Domain:
Record SIP history: Off
Call Events: On
Auth. Failure Events: Off
T.38 support: No
T.38 EC mode: Unknown
T.38 MaxDtgrm: 4294967295
SIP realtime: Disabled
Qualify Freq : 60000 ms
Q.850 Reason header: No
Store SIP_CAUSE: No
Network QoS Settings:
---------------------------
IP ToS SIP: CS3
IP ToS RTP audio: EF
IP ToS RTP video: AF41
IP ToS RTP text: CS0
802.1p CoS SIP: 4
802.1p CoS RTP audio: 5
802.1p CoS RTP video: 6
802.1p CoS RTP text: 5
Jitterbuffer enabled: No
Network Settings:
---------------------------
SIP address remapping: Disabled
Externhost: <none>
Externaddr: (null)
Externrefresh: 10
Localnet: 192.168.2.0/255.255.255.0
Global Signalling Settings:
---------------------------
Codecs: (gsm|ulaw|alaw|g726)
Codec Order: ulaw:20,alaw:20,gsm:20,g726:20
Relax DTMF: No
RFC2833 Compensation: No
Symmetric RTP: Yes
Compact SIP headers: No
RTP Keepalive: 0 (Disabled)
RTP Timeout: 30
RTP Hold Timeout: 300
MWI NOTIFY mime type: application/simple-message-summary
DNS SRV lookup: No
Pedantic SIP support: Yes
Reg. min duration 60 secs
Reg. max duration: 3600 secs
Reg. default duration: 120 secs
Sub. min duration 60 secs
Sub. max duration: 3600 secs
Outbound reg. timeout: 20 secs
Outbound reg. attempts: 0
Outbound reg. retry 403:0
Notify ringing state: Yes
Include CID: No
Notify hold state: Yes
SIP Transfer mode: open
Max Call Bitrate: 384 kbps
Auto-Framing: No
Outb. proxy: <not set>
Session Timers: Accept
Session Refresher: uas
Session Expires: 1800 secs
Session Min-SE: 90 secs
Timer T1: 500
Timer T1 minimum: 100
Timer B: 32000
No premature media: Yes
Max forwards: 70
Default Settings:
-----------------
Allowed transports: UDP
Outbound transport: UDP
Context: from-sip-external
Record on feature: automon
Record off feature: automon
Force rport: Yes
DTMF: rfc2833
Qualify: 0
Keepalive: 0
Use ClientCode: No
Progress inband: Never
Language:
Tone zone: <Not set>
MOH Interpret: default
MOH Suggest:
Voice Mail Extension: *97
----Name/username Host Dyn Forcerport Comedia ACL Port Status Description
1001/1001 1.39.63.239 D Yes Yes A 28594 UNREACHABLE
1002/1002 106.200.190.71 D Yes Yes A 47695 OK (216 ms) 这是我上次的会话。
这里的用户1001是"不可达"为什么?我想这就是我的问题所在。
帮帮我……
我也在寻找与PSTN和GSM连接的方法。
(如果你们来自印度，可以帮助我，我可以付钱给你，请回答上述问题的解决方案，然后我会联系其他方法)
您必须在sip.conf的[general]部分添加externip=your_public_ip。此外，您还必须转发RTP端口范围。通常是10000-20000 UDP。您可以在rtp.conf中看到/更改此范围。
当服务器处于Nat后时，SIP总是会导致问题。
如果你的设备支持IAX，这是星号间交换，完美地为您的情况，然后利用它。
您仍然希望解决SIP问题，请阅读本教程