爲什麼'頻道發起'呼叫？

我測試用：

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

PHONE是某人的電話號碼。

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

但手機不響。

注意事項：昨天有效。我可以使用這個命令來調用外部電話。但我無法從外面伸手。

今天，神奇的是，沒有改變我的配置，這是倒過來的。該命令不起作用。但我可以從外面伸手。

這裏是我的調試轉儲（從SIP組調試）

SIP_EXTENSION的電話號碼我叫到達我的上下文。

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) ---

你有什麼想法是什麼問題？

來源

2016-04-03 Pierre-Luc BLOT

最喜歡你的後備箱有不正確的設置。

欲瞭解更多信息行動之前做：

asterisk -r 
sip set debug on

來源

2016-04-04 00:09:19 arheops

我在問題中添加了我的調試轉儲;） –

調試是SO的功能。我可以推薦你與轉儲聯繫OVH。 – arheops

檢查尾的/ var /日誌/星號/消息。可能有多種原因。被叫號碼（PHONE）可能不會回答，或者您的SIP幹線是我們的信用。這些不在星號上下文中。但是當pbx（星號內部通道API）無法啓動時，這可能是一個本地問題。內存不足，在撥號方案中無法找到最大併發呼叫限制，上下文或擴展，並且其他一些事情可能導致此問題。

來源

2016-04-04 15:10:58 viktike

爲什麼'頻道發起'呼叫？

回答

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