Question

我想邀请某人加入一个环境。

我测试：

asterisk*CLI> channel originate SIP/trunk-test/PHONE extension s@my-context

PHONE是某人的电话号码。

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

命令打印：

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

但电话不响。

注意：昨天有效。我可以用这个命令来拨打外接电话。但我无法从外面接触到。

今天，奇迹般地，我的配置没有变化，这是颠倒的。此命令不起作用。但是我可以从外面接触到。

这是我的调试转储（来自sip set debug on）

SIP_EXTENSION是我拨打电话号码到达我的上下文。

asterisk*CLI> channel originate SIP/trunk-test/PHONE extension s@my-context
  == 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:PHONE@siptrunk.ovh.net SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK5158c2d4;rport
Max-Forwards: 70
From: "Anonymous" <sip:SIP_EXTENSION@siptrunk.ovh.net>;tag=as6a21d2d9
To: <sip:PHONE@siptrunk.ovh.net>
Contact: <sip:SIP_EXTENSION@192.168.1.10:5060>
Call-ID: 3dfff18c396337fb3c55a3fb7bc8c4c3@siptrunk.ovh.net
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:SIP_EXTENSION@siptrunk.ovh.net>;tag=as6a21d2d9
To: <sip:PHONE@siptrunk.ovh.net>
Call-ID: 3dfff18c396337fb3c55a3fb7bc8c4c3@siptrunk.ovh.net
CSeq: 102 INVITE
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---
Transmitting (NAT) to 192.168.1.1:5060:
ACK sip:PHONE@siptrunk.ovh.net SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK5158c2d4;rport
Max-Forwards: 70
From: "Anonymous" <sip:SIP_EXTENSION@siptrunk.ovh.net>;tag=as6a21d2d9
To: <sip:PHONE@siptrunk.ovh.net>
Contact: <sip:SIP_EXTENSION@192.168.1.10:5060>
Call-ID: 3dfff18c396337fb3c55a3fb7bc8c4c3@siptrunk.ovh.net
CSeq: 102 ACK
User-Agent: Asterisk PBX certified/13.1-cert4
Content-Length: 0


---
Scheduling destruction of SIP dialog '3dfff18c396337fb3c55a3fb7bc8c4c3@siptrunk.ovh.net' 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:SIP_EXTENSION@siptrunk.ovh.net>;tag=as6a21d2d9
To: <sip:PHONE@siptrunk.ovh.net>
Call-ID: 3dfff18c396337fb3c55a3fb7bc8c4c3@siptrunk.ovh.net
CSeq: 102 ACK
Content-Length: 0

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

您对这个问题有什么想法吗？

Answer 1

检查 tail / var / log / asterisk / messages 。可能有多种原因。被叫号码（PHONE）可能无法应答，或者您的SIP中继是我们的功劳。这些不在星号上下文中。但它可能是一个本地问题，这通常发生在无法启动pbx（星号内部通道API）时。内存不足，在拨号方案中找不到最大并发呼叫限制，上下文或扩展名，而其他一些内容可能会导致此问题。

Answer 2

最可能的是你的后备箱设置不正确。

如需了解更多信息，请先行动：

asterisk -r
sip set debug on

为什么没有＆＃39;频道来源＆＃39;呼叫？

2 个答案: