Asterisk 星号16 w/PJSIP转发入站呼叫时-“每个人都很忙/拥挤”

我在CentOS 7和PJSIP上运行星号16。我正在尝试将所有入站呼叫转接到由18001112222表示的电话号码，再转接到由12224446666表示的手机。向提供商注册和进入PBX的入站呼叫都可以正常工作，但创建到我手机的出站转发呼叫会导致以下日志行：

  == Setting global variable 'SIPDOMAIN' to '167.71.106.189'
    -- Executing [18001112222@from-external:1] Verbose("PJSIP/mytrunk-00000000", "1, "New caller, 2224446666 dialed into the IVR."") in new stack
  "New caller, 2224446666 dialed into the IVR."
    -- Executing [18001112222@from-external:2] Answer("PJSIP/mytrunk-00000000", "") in new stack
    -- Executing [18001112222@from-external:3] Dial("PJSIP/mytrunk-00000000", "PJSIP/mytrunk/sip:14045964207") in new stack
    -- Called PJSIP/mytrunk/sip:14045964207
    -- PJSIP/mytrunk-00000000 requested media update control 26, passing it to PJSIP/mytrunk-00000001
  == Everyone is busy/congested at this time (1:0/1/0)
    -- Executing [18001112222@from-external:4] Hangup("PJSIP/mytrunk-00000000", "") in new stack
  == Spawn extension (from-external, 18001112222, 4) exited non-zero on 'PJSIP/mytrunk-00000000'

此时每个人都很忙/拥挤，这是导致此设置失败的错误消息。我曾尝试使用许多在线论坛帖子更改mytrunk的身份验证方案，但错误消息仍然显示在转发的出站部分。我的配置文件如下：

pjsip.conf：

我认为问题在于出站调用的pjsip.conf配置，因为入站调用到达PBX时没有问题，我可以执行类似于Verbose的操作，并在Asterisk CLI中查看输出。当来自ITSP的入站呼叫成功时，我不确定要创建出站分支需要做什么

谢谢。

代替

same = n,Dial(PJSIP/mytrunk/sip:12224446666@mytrunk)

试一试

---

因为所有配置都在主干声明中，所以不需要将uri发送到主干。

本主题没有编程问题。这只是调试。您应该进行pjsip调试，并查看使用了哪个对等机。

[globals]
; General internal dialing options used in context Dial-Users.
; Only the timeout is defined here. See the Dial app documentation for
; additional options.
INTERNAL_DIAL_OPT=,30

[from-external]
exten = _18001112222,1,Verbose(1, "New caller, ${CALLERID(num)} dialed in.")
 same = n,Answer()
 same = n,Dial(PJSIP/mytrunk/sip:12224446666@mytrunk)
 same = n,Hangup()

same = n,Dial(PJSIP/mytrunk/sip:12224446666@mytrunk)

same = n,Dial(PJSIP/12224446666@mytrunk)