我已经从一个Asterisk(version 11.2.1)服务器创建了一个sip通道到另一个Asterisk服务器(11.7.0),称为'B',并且收到sip响应200 ok。
但是当我在Asterisk A上拨打DID时,通话被路由到Asterisk 'B',并且在38秒后,显示以下警告后通话已经断开:
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?
sip.conf
中的NAT设置造成的。 - Vivek Raj