创建两个邀请的星号拨号应用程序

问题描述

我在使用 Asterisk 13.38.2 时遇到问题，希望有人能帮助我。 当我的 Asterisk 收到一个 sip 邀请然后拨号到分机（如 Dial(SIP/103)）时，它按预期工作，只生成一个到分机 103 的邀请。

当相同的 DID 被路由到队列时，Asterisk 会创建两个 sip 邀请（一个用于预期的分机，另一个用于呼叫者联系地址）。

知道为什么创建最后一个邀请吗？

asterisk -rx "队列显示 600":

600 has 0 calls (max unlimited) in 'linear' strategy (0s holdtime,0s talktime),W:0,C:0,A:0,SL:0.0% within 60s
Members: 
      Local/103@test-queue (ringinuse enabled) (dynamic) (Not in use) has taken no calls yet
   No Callers

这是我的测试拨号计划：

# extension dial context
[test-queue]
exten => _X.,1,Dial(SIP/103)

# incoming context
[test-int]
exten => s,Answer()
same => n,Queue(600)

这些是邀请：

# Invite received by Asterisk
INVITE sip:[phone]@1.1.1.180:5060 SIP/2.0
Record-Route: <sip:1.1.1.18:5060;r2=on;lr;ftag=p65540t1617838888m773513c7005s1_2017909308-538403679>
Record-Route: <sip:1.1.1.18;transport=tcp;r2=on;lr;ftag=p65540t1617838888m773513c7005s1_2017909308-538403679>
Via: SIP/2.0/UDP 1.1.1.18:5060;branch=z9hG4bK4b8e.e54a976359c351d06657262d335cf2ea.0;i=1
Via: SIP/2.0/TCP 1.1.1.38:5060;branch=z9hG4bKkitt7g10305hb5b2mao0.1
To: <sip:[phone]@1.1.1.180;user=phone>
From: sip:[phone]@1.1.1.18;user=phone;tag=p65540t1617838888m773513c7005s1_2017909308-538403679
Call-ID: p65540t1617838888m773513c7005s2
CSeq: 1 INVITE
Max-Forwards: 65
Content-Length: 1060
Contact: <sip:p65540t1617838888m773513c7005s1@1.1.1.38:5060;transport=tcp>
Content-Type: application/sdp
Allow: REGISTER,REFER,NOTIFY,SUBSCRIBE,PRACK,UPDATE,INVITE,ACK,OPTIONS,CANCEL,BYE
Accept: application/sdp
Accept-Contact: xxxxxxxxxxxxxxxxx
Supported: timer,100rel,histinfo
P-Asserted-Identity: <sip:[phone]@1.1.1.38;cpc=ordinary>
History-Info: <sip:[phone]@ims2xxxxxxxx;user=phone>;index=1
History-Info: <sip:[phone]@ims2xxxxxxxx;cause=302;user=phone>;index=1.1
Min-SE: 900
Session-Expires: 1800
P-Charging-Vector: icid-value=mOTin606e4328d92c500100001b5d;icid-generated-at=ssdfd.xxxx.xxx.xx;orig-ioi=xxxx.xxx.xx;orig-icid=13212312131231
P-Early-Media: supported
Session-ID: a3a86bb42ac1e8c58592a9663bbd5ed5

# Invite sent to extension (correct)
INVITE sip:[e-mail];transport=ws SIP/2.0
Via: SIP/2.0/WS 1.1.1.180:5060;branch=z9hG4bK544ff0c7;rport
Max-Forwards: 70
From: "[phone]" <sip:[phone]@1.1.1.180>;tag=as185ed728
To: <sip:[e-mail];transport=ws>
Contact: <sip:[phone]@1.1.1.180:5060;transport=ws>
Call-ID: 5ece1d376d522a03461cb9671f6af1ec@1.1.1.180:5060
CSeq: 102 INVITE
Date: Wed,07 Apr 2021 23:41:33 GMT
Allow: INVITE,BYE,INFO,PUBLISH,MESSAGE
Supported: replaces,timer
Content-Type: application/sdp
Content-Length: 794

# No idea why Dial(SIP/103) created this invite!
INVITE sip:p65540t1617838888m773513c7005s1@1.1.1.38:5060;transport=tcp SIP/2.0
Via: SIP/2.0/UDP 1.1.1.180:5060;branch=z9hG4bK77194dbb
Route: <sip:1.1.1.18:5060;r2=on;lr;ftag=p65540t1617838888m773513c7005s1_2017909308-538403679>,<sip:1.1.1.18;transport=tcp;r2=on;lr;ftag=p65540t1617838888m773513c7005s1_2017909308-538403679>
Max-Forwards: 70
From: <sip:[phone]@1.1.1.180;user=phone>;tag=as7a03d243
To: sip:[phone]@1.1.1.18;user=phone;tag=p65540t1617838888m773513c7005s1_2017909308-538403679
Contact: <sip:[phone]@1.1.1.180:5060>
Call-ID: p65540t1617838888m773513c7005s2
CSeq: 102 INVITE
Session-Expires: 1800;refresher=uac
Min-SE: 90
Allow: INVITE,timer
P-Asserted-Identity: "Test" <sip:103@1.1.1.18>
Content-Type: application/sdp
Content-Length: 614

解决方法

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