星号，SIP 重传超时 [英] Asterisk,SIP Retransmission timeout
问题描述
我已经从一个 Asterisk(版本 11.2.1)创建了一个 sip 中继，说A"服务器到另一个 Asterisk 服务器(11.7.0)说B"，我得到 sip 响应 200 ok.
但是，当我开始呼叫 Asterisk A 上的 DID 时，呼叫被路由到 Asterisk 'B'，并且在 38 秒后呼叫已断开，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
当你遇到 nat 问题或防火墙问题时，这种情况就会出现
Such situation can be spot when you have nat issues or firewall issue
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
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！
