ckleea

How many servers you have now?

角色

问题： 用SPA IP Phone注册可以（例如ext 2001），但是在Asterisk Server看到ext 2001 unreachable

解答：
1） 我的第二台Asterisk server B全用default port 5060，而我利用我的router，做port redirection，从5228 UDP Port转到5060 UDP Port of Asterisk server B IP，于是我可以注册到。

2）但是这个“转动”改动是单向，而不是双向， 所以Asterisk server B能注册，在Asterisk server B想发出UDP packets to IP Phone时，因为router不会把Asterisk server B IP:5060, 往外面转至WAN （IP） Port：5228。

3）在更改Asterisk server B的bindport=5228，和router，用port forwarding，WAN的5228，转到Asterisk server IP的5228，就可以。这个转向是双向的。

4）更改后SPA IP Phone注册后，就可以reachable。


角色

角色

问题： 用5228 TCP Port注册不成功？

解答：
因为Asterisk default是用UDP port，从TCP更改到UDP port，变成注册。

角色

角色

回復 16# ckleea

Now I have two servers running at the same time，one is QNAP TS-119 NAS Asterisk and the other is Intel ATOM-based D515 Asterisk. I may add another one IP-01 later. If I have time, I may add one more for testing purpose.

YH

ckleea

I recall I do not need to make this port change when I have two asterisk servers. But I use two public IP.

角色

If you do have two public IP addresses, you do not need this port forwarding process.

YH

角色

问题： Asterisk Server B可以注册到，但是内外互打出问题，两边什么声音都听不到。

答案：
1) 我Asterisk Server A的rtp.conf已经更改为10000-10999, 而Server A，已经更改过。而Server B, router都做了port forwarding，如UDP 5228， 11000-11999，分流到Server B IP address。

2） 查找后，发现在router对应Server A的RTP port forwarding 10000-20000, 没有更新，现在更改为10000-10999 (因为没有时间测试，所以不知道这次更改后，是否通话正常？）

角色

ckleea

It should work for you

角色

It is confirmed that with the revised settings, both SIP signalling and RTP voice packets work perfectly.

YH

ttmuskie

Registered but no sound, my NAT problem?

1. Router set port Forward 7000-7999 (both TCP and UCP)
2. sip.conf

1. [general]
   
2. bindport=7060
   
3. [1234]
   
4. nat=yes
   
5. ...
   
6. ...

複製代碼

3. rtp.conf

1. rtpstart=7900
   
2. rtpend=7999
   
3. ...
   
4. ...

複製代碼

Is there anything I need to set? Thanks!

角色

Are you testing the Asterisk within the network segment?

What is the model number of QNAP that you using?

YH

ttmuskie

1. I'm testing it from outside and connect back to my home network
2. I don't have any QNAP and my asterisk is run under ubuntu.

I just figured out my Android SIP client (CSIPSimple) need to enable the stun server in order to hear the sound. Any recommendation for the stun server? Thanks.

bubblestar

It depends.  It is not necessary for me to enable stun in Asterisk and other voip devices behind NAT.  If you need to , you may try this one stun.xten.com