为什么'频道发起'呼叫？

我测试用：

asterisk*CLI> channel originate SIP/trunk-test/PHONE extension [email protected]

PHONE是某人的电话号码。

my-context is a context into my dialplan which contains a MeetMe Room

的命令打印：

== Using SIP RTP CoS mark 5 
    -- Called trunk-test/PHONE

但手机不响。

注意事项：昨天有效。我可以使用这个命令来调用外部电话。但我无法从外面伸手。

今天，神奇的是，没有改变我的配置，这是倒过来的。该命令不起作用。但我可以从外面伸手。

这里是我的调试转储（从SIP组调试）

SIP_EXTENSION的电话号码我叫到达我的上下文。

asterisk*CLI> channel originate SIP/trunk-test/PHONE extension [email protected] 
    == Using SIP RTP CoS mark 5 
Audio is at 15896 
Adding codec ulaw to SDP 
Adding codec alaw to SDP 
Adding codec gsm to SDP 
Reliably Transmitting (NAT) to 91.121.129.23:5060: 
INVITE sip:[email protected] SIP/2.0 
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK5158c2d4;rport 
Max-Forwards: 70 
From: "Anonymous" <sip:[email protected]>;tag=as6a21d2d9 
To: <sip:[email protected]> 
Contact: <sip:[email protected]:5060> 
Call-ID: [email protected] 
CSeq: 102 INVITE 
User-Agent: Asterisk PBX certified/13.1-cert4 
Date: Mon, 04 Apr 2016 01:34:56 GMT 
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE 
Supported: replaces, timer 
Content-Type: application/sdp 
Content-Length: 246 

v=0 
o=root 1530515320 1530515320 IN IP4 37.187.205.63 
s=Asterisk PBX certified/13.1-cert4 
c=IN IP4 MY_PUBLIC_IP 
t=0 0 
m=audio 15896 RTP/AVP 0 8 3 
a=rtpmap:0 PCMU/8000 
a=rtpmap:8 PCMA/8000 
a=rtpmap:3 GSM/8000 
a=maxptime:150 
a=sendrecv 

--- 
    -- Called trunk-test/PHONE 

<--- SIP read from UDP:192.168.1.1:5060 ---> 
SIP/2.0 408 Request Timeout 
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK5158c2d4;rport 
From: "Anonymous" <sip:[email protected]>;tag=as6a21d2d9 
To: <sip:[email protected]> 
Call-ID: [email protected] 
CSeq: 102 INVITE 
Content-Length: 0 

<-------------> 
--- (7 headers 0 lines) --- 
Transmitting (NAT) to 192.168.1.1:5060: 
ACK sip:[email protected] SIP/2.0 
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK5158c2d4;rport 
Max-Forwards: 70 
From: "Anonymous" <sip:[email protected]>;tag=as6a21d2d9 
To: <sip:[email protected]> 
Contact: <sip:[email protected]:5060> 
Call-ID: [email protected] 
CSeq: 102 ACK 
User-Agent: Asterisk PBX certified/13.1-cert4 
Content-Length: 0 


--- 
Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: INVITE) 

<--- SIP read from UDP:192.168.1.1:5060 ---> 
SIP/2.0 408 Request Timeout 
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK5158c2d4;rport 
From: "Anonymous" <sip:[email protected]>;tag=as6a21d2d9 
To: <sip:[email protected]> 
Call-ID: [email protected] 
CSeq: 102 ACK 
Content-Length: 0 

<-------------> 
--- (7 headers 0 lines) ---

你有什么想法是什么问题？

来源

2016-04-03 Pierre-Luc BLOT

最喜欢你的后备箱有不正确的设置。

欲了解更多信息行动之前做：

asterisk -r 
sip set debug on

来源

2016-04-04 00:09:19 arheops

我在问题中添加了我的调试转储;） –

调试是SO的功能。我可以推荐你与转储联系OVH。 – arheops

检查尾的/ var /日志/星号/消息。可能有多种原因。被叫号码（PHONE）可能不会回答，或者您的SIP干线是我们的信用。这些不在星号上下文中。但是当pbx（星号内部通道API）无法启动时，这可能是一个本地问题。内存不足，在拨号方案中无法找到最大并发呼叫限制，上下文或扩展，并且其他一些事情可能导致此问题。

来源

2016-04-04 15:10:58 viktike

为什么'频道发起'呼叫？

回答

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