我有一台FreeSwitch服务器（Ubuntu上的1.4.26）。 在连接呼叫后30分钟将来电重定向到外部服务器时，我从目标服务器收到RE-INVITE消息。我的FreeSwitch服务器响应“481呼叫不存在”，然后呼叫被断开，虽然它很顺利。
我假设在“Session-Expires：3600; refresher = uac”的一半时间后发送了RE-INVITE。
我试图告诉FreeSwitch忽略重新邀请，在桥接之前使用set sip_ignore_reinvites = true。似乎没有任何影响。也试过桥的起源字符串。没有帮助。
如何防止这种情况发生？
以下是SIP日志（1111呼叫9999）：
send 1069 bytes to udp/[99.99.99.99]:5060 at 15:02:29.531004:
------------------------------------------------------------------------
INVITE sip:12129999999@99.99.99.99:5060 SIP/2.0
Via: SIP/2.0/UDP 55.55.55.55;rport;branch=z9hG4bKvjUSX912pUc9Q
Max-Forwards: 67
From: "12121111111" <sip:12121111111@55.55.55.55>;tag=p02B1veKSg0tS
To: <sip:12129999999@99.99.99.99:5060>
Call-ID: ATUCMSOAWBAMZGOJMMLEQ7CU74@11.11.11.48_01
CSeq: 92276610 INVITE
Contact: <sip:mod_sofia@55.55.55.55:5060>
User-Agent: FreeSWITCH-mod_sofia/1.4.26+git~20151215T154940Z~5579a4ba46~64bit
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 223
X-FS-Support: update_display,send_info
Remote-Party-ID: "12121111111" <sip:12121111111@55.55.55.55>;party=calling;screen=yes;privacy=off
v=0
o=FreeSWITCH 1465208343 1465208344 IN IP4 55.55.55.55
s=FreeSWITCH
c=IN IP4 55.55.55.55
t=0 0
m=audio 17006 RTP/AVP 0 101 13
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
------------------------------------------------------------------------
recv 802 bytes from udp/[99.99.99.99]:5060 at 15:02:50.184437:
------------------------------------------------------------------------
SIP/2.0 200 OK
Via: SIP/2.0/UDP 55.55.55.55;received=55.55.55.55;branch=z9hG4bKvjUSX912pUc9Q;rport=5060
From: "12121111111" <sip:12121111111@55.55.55.55>;tag=p02B1veKSg0tS
To: <sip:12129999999@99.99.99.99:5060>;tag=9307555045152742154
Call-ID: ATUCMSOAWBAMZGOJMMLEQ7CU74@11.11.11.48_01
CSeq: 92276610 INVITE
Content-Type: application/sdp
Session-Expires: 3600;refresher=uas
Contact: <sip:12129999999@99.99.99.99:5060;user=phone;transport=udp>
Supported: timer,100rel
Content-Length: 288
v=0
o=MG4000|2.0 193121 196925 IN IP4 99.99.99.99
s=-
c=IN IP4 99.99.99.99
t=0 0
m=audio 61308 RTP/AVP 0 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=X-vrzcap:vbd Ver=0 Mode=FaxPr ModemRtpRed=0
a=X-vrzcap:identification bin=DSR2883 Prot=mgcp App=MG
------------------------------------------------------------------------
recv 934 bytes from udp/[99.99.99.99]:5060 at 15:32:50.190171:
------------------------------------------------------------------------
INVITE sip:mod_sofia@55.55.55.55:5060 SIP/2.0
Via: SIP/2.0/UDP 99.99.99.99:5060;branch=z9hG4bKi8jups3010shhe7um2g0sb0000g00.1
Call-ID: ATUCMSOAWBAMZGOJMMLEQ7CU74@11.11.11.48_01
From: <sip:12129999999@99.99.99.99:5060>;tag=9307555045152742154
To: "12121111111" <sip:12121111111@55.55.55.55>;tag=p02B1veKSg0tS
Content-Type: application/sdp
Min-SE: 90
Session-Expires: 3600;refresher=uac
CSeq: 1 INVITE
Contact: <sip:12129999999@99.99.99.99:5060;user=phone;transport=udp>
Allow: INVITE,CANCEL,BYE,ACK,REFER,UPDATE,INFO,PRACK
Supported: timer,100rel
Max-Forwards: 69
User-Agent: VCS 5.10.2.10-02
Content-Length: 288
v=0
o=MG4000|2.0 193121 196925 IN IP4 99.99.99.99
s=-
c=IN IP4 99.99.99.99
t=0 0
m=audio 61308 RTP/AVP 0 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=X-vrzcap:vbd Ver=0 Mode=FaxPr ModemRtpRed=0
a=X-vrzcap:identification bin=DSR2883 Prot=mgcp App=MG
------------------------------------------------------------------------
send 513 bytes to udp/[99.99.99.99]:5060 at 15:32:50.190379:
------------------------------------------------------------------------
SIP/2.0 481 Call Does Not Exist
Via: SIP/2.0/UDP 99.99.99.99:5060;branch=z9hG4bKi8jups3010shhe7um2g0sb0000g00.1
From: <sip:12129999999@99.99.99.99:5060>;tag=9307555045152742154
To: "12121111111" <sip:12121111111@55.55.55.55>;tag=p02B1veKSg0tS
Call-ID: ATUCMSOAWBAMZGOJMMLEQ7CU74@11.11.11.48_01
CSeq: 1 INVITE
User-Agent:FreeSWITCH-mod_sofia/1.4.26+git~20151215T154940Z~5579a4ba46~64bit
Allow:INVITE,ACK,BYE,CANCEL,OPTIONS,MESSAGE,INFO,UPDATE,REGISTER,REFER,NOTIFY
k:timer,path,replaces
l:0
------------------------------------------------------------------------
recv 374 bytes from udp/[99.99.99.99]:5060 at 15:32:50.276999:
------------------------------------------------------------------------
ACK sip:mod_sofia@55.55.55.55:5060 SIP/2.0
Via: SIP/2.0/UDP 99.99.99.99:5060;branch=z9hG4bKi8jups3010shhe7um2g0sb0000g00.1
CSeq: 1 ACK
Call-ID: ATUCMSOAWBAMZGOJMMLEQ7CU74@11.11.11.48_01
From: <sip:12129999999@99.99.99.99:5060>;tag=9307555045152742154
To: "12121111111" <sip:12121111111@55.55.55.55>;tag=p02B1veKSg0tS
Max-Forwards: 69
Content-Length: 0
------------------------------------------------------------------------
recv 477 bytes from udp/[99.99.99.99]:5060 at 15:32:50.290275:
------------------------------------------------------------------------
BYE sip:mod_sofia@55.55.55.55:5060 SIP/2.0
Via: SIP/2.0/UDP 99.99.99.99:5060;branch=z9hG4bKi8jups3010shhe7um2g0sd0000010.1
Call-ID: ATUCMSOAWBAMZGOJMMLEQ7CU74@11.11.11.48_01
From: <sip:12129999999@99.99.99.99:5060>;tag=9307555045152742154
To: "12121111111" <sip:12121111111@55.55.55.55>;tag=p02B1veKSg0tS
CSeq: 2 BYE
Supported: timer,100rel
Max-Forwards: 69
Reason: SIP;cause=0;iintcode=516;isubsystem=0
User-Agent: VCS 5.10.2.10-02
Content-Length: 0
------------------------------------------------------------------------
send 510 bytes to udp/[99.99.99.99]:5060 at 15:32:50.290421:
------------------------------------------------------------------------
SIP/2.0 481 Call Does Not Exist
Via: SIP/2.0/UDP 99.99.99.99:5060;branch=z9hG4bKi8jups3010shhe7um2g0sd0000010.1
From: <sip:12129999999@99.99.99.99:5060>;tag=9307555045152742154
To: "12121111111" <sip:12121111111@55.55.55.55>;tag=p02B1veKSg0tS
Call-ID: ATUCMSOAWBAMZGOJMMLEQ7CU74@11.11.11.48_01
CSeq: 2 BYE
User-Agent:FreeSWITCH-mod_sofia/1.4.26+git~20151215T154940Z~5579a4ba46~64bit
Allow:INVITE,ACK,BYE,CANCEL,OPTIONS,MESSAGE,INFO,UPDATE,REGISTER,REFER,NOTIFY
k:timer,path,replaces
l:0
答案 0 :(得分：1)
如果以上是整个跟踪，那么现在在200之后从FS发送ACK是很奇怪的。重新邀请就像你说的会话刷新一样，在同一个对话框上发生，但它是一个不同的事务。
查看来自-tag / to-tag的呼叫ID，重新邀请看起来是正确的。
创建一个tcpdump / wireshark并确保重新邀请被发送到正确的端口，并且在初始邀请200 ok之后有一个ACK
答案 1 :(得分：1)
您是否尝试在Freeswitch上启用RFC 4028支持？
https://wiki.freeswitch.org/wiki/Sofia.conf.xml#SIP_Related_options
在您的sip个人资料中：
<param name="enable-timer" value="true"/>
答案 2 :(得分：0)
您应该在重新邀请中删除to标记（tag = p02B1veKSg0tS），因为这是一个新事务。