星号拨打电话警告纯数字主机名

时间:2019-06-19 11:13:52

标签: call asterisk out

我安装了FreePbx。但是我有一个问题。我想打自己的手机,但不能。

我有一个IP电话。我将他的类型从pjsip更改为sip。

扩展名:102

extensions.conf:

[cocugunuzukarsilayin]
exten => myphonenumber,1,Answer()
exten => myphonenumber,n,Wait(1)
exten => myphonenumber,n,Playback(custom/sound2)
exten => myphonenumber,n,Wait(1)
exten => myphonenumber,n,Hangup()

test.call

Channel: SIP/102/myphonenumber
MaxRetries: 2
RetryTime: 30
WaitTime: 15
Context: cocugunuzukarsilayin
Extension: cocugunuzukarsilayin
Priority: 2

错误:

[2019-06-19 14:08:39] WARNING[8744]: chan_sip.c:6274 create_addr: Purely numeric hostname (102), and not a peer--rejecting!
[2019-06-19 14:08:39] NOTICE[8744]: pbx_spool.c:447 attempt_thread: Call failed to go through, reason (0) Call Failure (not BUSY, and not NO_ANSWER, maybe Circuit busy or down?)
[2019-06-19 14:08:39] WARNING[8744]: pbx_spool.c:350 safe_append: Unable to set utime on /var/spool/asterisk/outgoing/test.call: Operation not permitted

出什么问题了?

2 个答案:

答案 0 :(得分:0)

频道格式如下:

...
  kafka-node:KafkaClient kafka-node-client reconnecting to ADDR:9092 +1s
  kafka-node:KafkaClient kafka-node-client createBroker ADDR:9092 +2ms
  kafka-node:Consumer connection closed +1s
  kafka-node:KafkaClient kafka-node-client socket closed ADDR:9092 (hadError: true) +3ms
...

请检查提到的'102'是有效的网关。

答案 1 :(得分:0)

我有同样的错误信息。而且,我无法拨打电话。我的消息日志具有以下顺序:

chan_sip.c: Purely numeric hostname (1001), and not a peer--rejecting!
app_dial.c: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)

通过阅读here,我终于找到了解决方案。我的问题是由 extensions.conf 中的技术类型不匹配引起的。我正在从 sip.conf 切换到 pjsip.conf 的过程中。但是我无法正确更新 extensions.conf

exten => 1001,1,Dial(SIP/1001,20,Ttm)
same  => n,Hangup

解决方法如下:

exten => 1001,1,Dial(PJSIP/1001,20,Ttm)
same  => n,Hangup

SIP 更改为 PJSIP 后,错误消息将被消除,并且恢复了正常功能。