Asmaa Ahmed wrote: > > > 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, a nd lead the call to be disconnected after! By checking the > logs, I can see this chan_sip.c:3641 retrans_pkt: Retransmission > timeout reached on transmission > Mjk3MGU1NjgxZWQwM2E3MjhjZmFiNzhjOGVjZjg5ZTc for seqno 2 > (Critical Response)
The SIP trace you provided breaks down as follows: X-Lite Asterisk --------------- ------------------------------- INVITE(No Auth) ---> <--- 401 Unauthorized ACK ---> INVITE(Auth) ---> <--- 100 Trying <--- 200 OK <--- 200 OK (Retransmitted 10 Times) <--- BYE OK ---> This shows that the three-way handshake (INVITE/200 OK/ACK) used to establish SIP sessions is not completed because Asterisk never receives an ACK from X-Lite. After retransmitting the 200 OK 10 times Asterisk gives up and disconnects the call. > Here's my simple sip configuration > [general] > context=internal > allowguest=no > allowoverlap=no > bindport=5060 > bindaddr=0.0.0.0 > srvlookup=no > disallow=all > allow=ulaw > alwaysauthreject=yes > canreinvite=no > nat=yes > session-timers=refuse > externip=<IP> >From the SIP trace, I believe 'externip=41.46.164.96' is set. If that is the case, try changing it to 'externip=54.241.129.14'. You should also set localnet as follows: ; RFC 1918 addresses localnet=192.168.0.0/255.255.0.0 localnet=10.0.0.0/255.0.0.0 localnet=172.16.0.0/12 If that doesn't work you can also try setting 'nat=force_rport' instead of 'nat=yes'. > [7001] > type=friend > host=dynamic > secret=123 > context=internal > > [7002] > type=friend > host=dynamic > secret=456 > context=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 > At this point, you should be able to establish a call between the two X-Lite phones that won't get disconnected due to failing to complete the three-way handshake. There may still not be voice because the firewall(s) between Asterisk and the X-Lite phones may block the RTP traffic. The phones appear to be on the same network, so you can try setting 'canreinvite=yes' to workaround this problem until the firewall(s) are configured to allow RTP traffic on the UDP port range specified in 'rtp.conf' (the default range is 10000-20000). Good luck and please report your progress back to the list. Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users