Hello, I am trying to make my first call on Asterisk to succeed. I have Asterisk 1.8.10.1 running on Ubuntu machine.The configuration is quite simple just for my first test, Trying to have a call between two X-lite sipphone. The subscribers succeeded to register and the call is established, but still no voice can be heard, and lead the call to be disconnected after! By checking the logs, I can see thischan_sip.c:3641 retrans_pkt: Retransmission timeout reached on transmission Mjk3MGU1NjgxZWQwM2E3MjhjZmFiNzhjOGVjZjg5ZTc for seqno 2 (Critical Response) Here's my simple sip configuration[general]context=internalallowguest=noallowoverlap=nobindport=5060bindaddr=0.0.0.0srvlookup=nodisallow=allallow=ulawalwaysauthreject=yescanreinvite=nonat=yessession-timers=refuseexternip=<IP>[7001]type=friendhost=dynamicsecret=123context=internal[7002]type=friendhost=dynamicsecret=456context=internal A snoop capture for my call is uploaded in the following link. I wonder if there is any missing configuration or plugin need to be set here!http://www.fileconvoy.com/dfl.php?id=gc0957418962ed157999374318118ae80b9d015992 Thanks.
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