星号，SIP重传超时 [英] Asterisk,SIP Retransmission timeout
问题描述
我已经创建了一个从Asterisk(版本11.2.1)说"A"服务器到另一个Asterisk服务器(11.7.0)说"B"的Sip中继，并且我得到了200 sip响应.
但是，当我开始在Asterisk A上调用DID时，该呼叫将路由到Asterisk'B'，并且在38秒后，呼叫已断开，并显示以下警告:
I have created a sip trunk from One Asterisk(version 11.2.1) say 'A' server to another Asterisk server(11.7.0) say 'B', and I am getting sip response 200 ok.
But when I start calling on a DID on Asterisk A then the call is being routed to Asterisk 'B' and After 38 seconds call has been disconnected showing following warnings :
Retransmission timeout reached on transmission 11bc71e029119e5877806ed40fcde691@111.xxx.xxx.xxx:5060 for seqno 102 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32000ms with no response
Hanging up call 11bc71e029119e5877806ed40fcde691@111.xxx.xxx.xx:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
有什么想法吗?
推荐答案
当您遇到nat问题或防火墙问题时，可能会发现这种情况
Such situation can be spot when you have nat issues or firewall issue
查看这篇文章 http://www.voip-info.org/wiki/view/Asterisk + SIP + NAT +解决方案
有关更多信息，您可以使用启用sip调试
For more info you can enable sip debug by using
asterisk -r
sip set debug on
这篇关于星号，SIP重传超时的文章就介绍到这了，希望我们推荐的答案对大家有所帮助，也希望大家多多支持IT屋！
