Check here: http://www.freepbx.org/forum/freepbx/installation/incoming-calls-drop-30-seconds-on-fpbx-2-9-0-1-8-7-0
Thanks, Steve Totaro On Mon, Mar 10, 2014 at 4:43 PM, alp...@gmail.com <alp...@gmail.com> wrote: > Yes, well, really is Elastix. Hmmm where I need to pt directmedia=no ? > > Thanks, > > > On Mon, Mar 10, 2014 at 2:38 PM, Eric Wieling <ewiel...@nyigc.com> wrote: > >> Try ulaw instead of g729, set directmedia=no >> >> I see you are using FreePBX. I cannot help further. >> >> >> -----Original Message----- >> From: asterisk-users-boun...@lists.digium.com [mailto: >> asterisk-users-boun...@lists.digium.com] On Behalf Of alp...@gmail.com >> Sent: Monday, March 10, 2014 4:15 PM >> To: Asterisk Users Mailing List - Non-Commercial Discussion >> Cc: and...@telesip.net >> Subject: Re: [asterisk-users] Remote extensions call drops after 20 >> seconds. >> >> Guys, hi. I have not solved the problem. Outgoing calls to remote >> extensions drops on 5-20 seconds. Incoming calls work perfectly. >> >> Hope you can help me. Attach updated logs: http://pastebin.com/HmKEDAUq >> >> Thanks, >> >> >> On Thu, Dec 19, 2013 at 8:48 AM, Eric Wieling <ewiel...@nyigc.com> wrote: >> >> >> See sip.conf.sample in the Asterisk tarball for documentation of >> valid settings. >> >> >> -----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 9:30 PM >> To: and...@telesip.net; Asterisk Users Mailing List - >> Non-Commercial Discussion >> Subject: Re: [asterisk-users] Remote extensions call drops after >> 20 seconds. >> >> >> I set canreinvite=very in the remote extension, and now the call >> not drops. Valid solution? >> >> >> On Wed, Dec 18, 2013 at 6:38 PM, Andres <and...@telesip.net> >> wrote: >> >> >> On 12/18/13, 3:09 PM, alp...@gmail.com wrote: >> >> >> 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 >> >> >> When the call is setup I see your Asterisk retransmitting >> the "SIP/2.0 200 OK" packet many times and getting no response. The other >> end needs to receive the packet and generate an "ACK". You need to trace >> where that packet is going and figure out why it is not reaching its >> target, or if it is, then why is the ACK not making it back. Thats your >> problem. >> >> >> Thank you! >> >> -- >> >> Allan Porras >> >> http://allanPorras.com < >> http://www.AllanPorras.com> >> Google Plus: http://goo.gl/BRkbX >> >> Twitter: @alpocr <http://twitter/alpocr> >> >> >> >> >> >> >> >> >> >> >> -- >> Technical Support >> http://www.cellroute.net >> >> -- >> >> _____________________________________________________________________ >> -- 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 >> >> >> >> >> >> -- >> >> 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