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>
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