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