here's a checklist... First, RTP port range not port forwarded correctly on the NAT router (check rtp.conf).
Then, on sip.conf: externip not correctly setup (it should be the public IP of the NAT router)? nat setting not enabled for any outbound trunk and the extensions (nat=yes) ? localnet not properly setup (to include subnets of local, un-nat'd extensions) ? canreinvite not disabled for any outbound trunk and for the extensions? rgds On Wed, Dec 18, 2013 at 8:34 PM, [email protected] <[email protected]> wrote: > Thank you Eric for your reply. How Can I fix it? > > In server side, I opened RTP ports. > > > On Wednesday, December 18, 2013, Eric Wieling wrote: > >> Calls dropping after 20 seconds is often directmedia enabled when it >> should not be enabled or RTP keepalives enabled when they should not be >> enabled. Dropping around 20 mins is often Session Timers being enabled >> when they don't work for the specific environment. >> >> -----Original Message----- >> From: [email protected] [mailto: >> [email protected]] On Behalf Of [email protected] >> Sent: Wednesday, December 18, 2013 3:09 PM >> To: [email protected] >> Subject: [asterisk-users] Remote extensions call drops after 20 seconds. >> >> Hello. I have a problem with the configuration of a remote extensions. >> Calls are truncated at 20 seconds. >> >> I got my my NAT firewall properly configured. Here I attached my debug in >> CLI: http://pastebin.com/gh34E69f >> >> Thank you! >> >> -- >> >> Allan Porras >> http://allanPorras.com <http://www.AllanPorras.com> Google Plus: >> http://goo.gl/BRkbX >> >> Twitter: @alpocr <http://twitter/alpocr> >> >> >> >> -- >> _____________________________________________________________________ >> -- 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 >> > > > -- > Allan Porras > http://allanPorras.com <http://www.AllanPorras.com> > Google Plus: http://goo.gl/BRkbX > Twitter: @alpocr <http://twitter/alpocr> > > > > > -- > _____________________________________________________________________ > -- 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 >
-- _____________________________________________________________________ -- 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
