Choose suitable NAT settings from sip.conf turn direct media in sip.conf or per peer off
On Thu, Sep 19, 2013 at 12:54 PM, Asmaa Ahmed <asabatg...@hotmail.com>wrote: > 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 this > chan_sip.c:3641 retrans_pkt: Retransmission timeout reached on > transmission Mjk3MGU1NjgxZWQwM2E3MjhjZmFiNzhjOGVjZjg5ZTc for seqno 2 > (Critical Response) > > 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> > > [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 > > <http://www.fileconvoy.com/dfl.php?id=gc0957418962ed157999374318118ae80b9d015992> > Thanks. > > > -- > _____________________________________________________________________ > -- 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 > -- Regards ************************** Muhammad Salman ***************************
-- _____________________________________________________________________ -- 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