带有PJSIP的星号16-转发呼入呼叫时，“每个人都忙/拥挤”

Question

我正在CentOS 7和PJSIP上运行Asterisk 16。我正在尝试将所有入站呼叫转移到电话号码（由12224446666代表）的电话号码（由18001112222代表）。向提供商进行注册和到达PBX的入站呼叫都可以正常工作，但是创建到我的手机的出站，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'

似乎Everyone is busy/congested at this time是导致此安装失败的错误消息。我尝试使用许多在线论坛帖子来更改mytrunk的身份验证方案，但是错误消息继续显示在转发者的出站分支上。我的配置文件如下：

pjsip.conf：

;==============TRANSPORTS

[simpletrans]
type=transport
protocol=udp
bind=0.0.0.0

;===============TRUNK

[mytrunk]
type=registration
outbound_auth=mytrunk
server_uri=sip:<SIP provider gateway>
client_uri=sip:<username>@<SIP provider gateway>
retry_interval=60

[mytrunk]
type=auth
auth_type=userpass
password=<password>
username=<username>

[mytrunk]
type=aor
contact=sip:<SIP provider gateway>:5060

[mytrunk]
type=endpoint
context=from-external
disallow=all
allow=ulaw
outbound_auth=mytrunk
aors=mytrunk

[mytrunk]
type=identify
endpoint=mytrunk
match=<SIP provider gateway>

extensions.conf：

[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()

我相信问题出在呼出电话]的pjsip.conf配置上，因为呼入电话无问题地到达了PBX，我可以执行类似Verbose()的操作并在Asterisk CLI中查看输出。当ITSP的入站呼叫成功时，我不确定创建出站支路需要做什么。

谢谢。

我正在CentOS 7和PJSIP上运行Asterisk 16。我正在尝试将所有入站呼叫转移到电话号码（由12224446666代表）的电话号码（由18001112222代表）。注册...

Answer 1

代替