Rodrigo, thanks for reply. 1- RTP ports is forwarded correctly on the NAT router. 2- externip is my public ip. 3- All my extensions have nat=yes by default. 4- localnet is setup. 5- canreinvite is disabled.
It could be a codec mistake? On Wed, Dec 18, 2013 at 2:58 PM, Rodrigo Borges Pereira < rodrigoborgespere...@gmail.com> wrote: > 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, alp...@gmail.com <alp...@gmail.com>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: asterisk-users-boun...@lists.digium.com [mailto: >>> asterisk-users-boun...@lists.digium.com] On Behalf Of alp...@gmail.com >>> Sent: Wednesday, December 18, 2013 3:09 PM >>> To: asterisk-users@lists.digium.com >>> 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 > -- 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