Hi guys, the problem was too many NATs on the way. Although the server had a valid ip, it was behind a nat, as soon as I set ip directly on the server, things worked fine. Also, despite the huge delay, if the link has qos, the quality is very good.
On Mon, Jan 16, 2012 at 9:06 AM, Sammy Govind <[email protected]> wrote: > I'm only expecting NAT issues if not the latency issues. SIP traces of any > such calls will make more sense. > > > On Mon, Jan 16, 2012 at 6:05 PM, Arlen Nascimento < > [email protected]> wrote: > >> the client is aware of the adverse environment and this is the only >> solution for him >> >> >> On Mon, Jan 16, 2012 at 9:00 AM, Flavio Miranda < >> [email protected]> wrote: >> >>> Unless you are doing test with SIP under adverse environmet, that is >>> not the point, but, if you intend to have Communication, you should worry >>> about this detail. >>> Basic infra-estructure is the first thing to think in any new project. >>> >>> Good luck! >>> >>> Att, >>> >>> Flavio Roberto Miranda >>> MSN:[email protected] >>> Skype: flaviormiranda >>> >>> ------------------------------ >>> Date: Mon, 16 Jan 2012 07:58:34 -0400 >>> From: [email protected] >>> To: [email protected] >>> Subject: Re: [asterisk-users] Peer doesn't answer >>> >>> >>> It is a satellite connection, so ping is about 500ms. I know it is not >>> ok to keep a normal conversation, that is not the point. >>> >>> >>> On Sun, Jan 15, 2012 at 10:34 PM, Flavio Miranda < >>> [email protected]> wrote: >>> >>> Hi Arlen, >>> >>> A reasonable time to Voip calls is about 250 ms. What about the Ping >>> test end-to-end ? >>> >>> Att, >>> >>> Flavio Roberto Miranda >>> MSN:[email protected] >>> Skype: flaviormiranda >>> >>> ------------------------------ >>> Date: Sun, 15 Jan 2012 21:53:46 -0400 >>> From: [email protected] >>> To: [email protected] >>> Subject: [asterisk-users] Peer doesn't answer >>> >>> >>> Hi all, >>> >>> i'm implementing an asterisk server that will have several peers >>> connected by satellite links. >>> When qualify=yes or some value (from 3000 to 50000), 'sip show peers' >>> shows the peer as unreachable. In this case i can place calls from the >>> phone in the satellite link, but can't call to it. >>> When i turn off qualify, the status changes to unmonitored. In this >>> case, I can make calls in both directions but the call is never >>> established. The phone keeps ringing until 'ring time' expires even when I >>> answer the call on the phone/softphone. >>> >>> Any thoughts? >>> >>> Regards, >>> >>> -- >>> Arlen Nascimento >>> >>> >>> -- _____________________________________________________________________ >>> -- 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 >>> >>> >>> >>> >>> -- >>> Arlen Nascimento >>> >>> >>> -- _____________________________________________________________________ >>> -- 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 >>> >> >> >> >> -- >> Arlen Nascimento >> >> >> -- >> _____________________________________________________________________ >> -- 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 > -- Arlen Nascimento
-- _____________________________________________________________________ -- 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
