为什么我的软电话无法连接到 Asterisk？

Question

我正在熟悉 Asterisk，并且正在阅读“Asterisk：电话的未来”一书。

我到那里我已经配置了点sip.conf，并extensions.conf根据书。

---

sip.conf：

[general]
context=default
allowoverlap=no
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes

[1000]
type=friend
context=phones
host=dynamic

---

extensions.conf：

[globals]

[general]
autofallthrough=yes

[default]
exten => s,1,Verbose(1|Unrouted call handler)
exten => s,n,Answer()
exten => s,n,Wait(1)
exten => s,n,Playback(tt-weasels)
exten => s,n,Hangup()

[incoming_calls]

[internal]
exten => 500,1,Verbose(1|Echo test application)
exten => 500,n,Echo()
exten => 500,n,Hangup()

[phones]
include => internal

---

我的 Asterisk 服务器和我的软电话 (Ekiga) 都在我的桌面上运行。当我添加一个帐户时，它说Could not register sip:1000@localhost。我还尝试将 Ekiga 中的注册商设置为127.0.0.1.

我只是想让 Asterisk 使用基本设置，这样我就可以试验它。

---

系统详情：

blaine :: /etc/asterisk » uname -a
Linux blaine 2.6.32-24-generic #38-Ubuntu SMP Mon Jul 5 09:20:59 UTC 2010 x86_64 GNU/Linux

blaine :: /etc/asterisk » cat /etc/issue
Ubuntu 10.04.1 LTS

---

更新

我在本地主机上运行了一个 nmap 并发现了以下内容

blaine :: /etc/asterisk » nmap localhost

Starting Nmap 5.00 ( http://nmap.org ) at 2010-08-04 10:38 MDT
Warning: Hostname localhost resolves to 2 IPs. Using 127.0.0.1.
Interesting ports on localhost (127.0.0.1):
Not shown: 987 closed ports
PORT     STATE SERVICE
22/tcp   open  ssh
80/tcp   open  http
631/tcp  open  ipp
1720/tcp open  H.323/Q.931
2000/tcp open  callbook
3306/tcp open  mysql
5222/tcp open  unknown
5269/tcp open  unknown
7070/tcp open  realserver
7443/tcp open  unknown
7777/tcp open  unknown
9090/tcp open  zeus-admin
9091/tcp open  unknown

Nmap done: 1 IP address (1 host up) scanned in 0.35 seconds

Answer 1

我相信我已经想通了。由于 SIP 客户端和服务器都在端口 5060 上运行，因此我不能在与我的软电话相同的设备上运行 Asterisk 服务器。我已经在远程服务器上安装了 Asterisk，我可以按照书中的说明使用 Ekiga 连接到它。

哈，书上也是这么说的。为我感到羞耻。

If you are running Asterisk and a softphone on the same system (i.e.,
running an X-Lite softphone and Asterisk on a laptop or desktop), then
you will need to modify the SIP port that client listens on. It will need
to be changed from 5060 to 5061 (or some other unused port) so that
Asterisk and the softphone do not interfere with each other.