Ok. I do NOT have ports 10000-20000 opened in. I guess I should try that and see if it works.
I will open ports 5060 - 5070 and 10000 - 100100 and do some test tonight. I will keep you posted. Thanks. --- On Wed, 11/18/09, Danny Nicholas <[email protected]> wrote: > From: Danny Nicholas <[email protected]> > Subject: Re: [asterisk-users] can't call through voip provider > To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" > <[email protected]> > Date: Wednesday, November 18, 2009, 5:18 PM > According to what I know, you have to > have 5060 open out and 10000-20000 > open in (you can cut this to as small as 10000-10004). > > -----Original Message----- > From: [email protected] > [mailto:[email protected]] > On Behalf Of Landy Landy > Sent: Wednesday, November 18, 2009 4:13 PM > To: Asterisk Users Mailing List - Non-Commercial > Discussion > Subject: Re: [asterisk-users] can't call through voip > provider > > According to the provider he says he doesn't see anything > coming in on their > side. I've had all ports FORWARD out to ACCEPT but, > blocking incoming new > connections. I thought when asterisk starts a communication > with a remote > server using an unprivate port to port 5060 theres already > an ESTABLISHED > communication. I don't know if I'm having problems with my > firewall script > or what but, since there isn't any new connections coming > form outside I > think I'm ok to accept only ESTABLISHED,RELATED coming in. > > I don't know but, I'm stuck with this problem and don't > know what else to > do. > > --- On Wed, 11/18/09, Warren Selby <[email protected]> > wrote: > > > From: Warren Selby <[email protected]> > > Subject: Re: [asterisk-users] can't call through voip > provider > > To: "Asterisk Users Mailing List - Non-Commercial > Discussion" > <[email protected]> > > Date: Wednesday, November 18, 2009, 5:03 PM > > What does your provider see when you > > attempt to call them? > > > > > > > > Thanks, > > --Warren Selby > > > > On Nov 18, 2009, at 3:38 PM, Landy Landy <[email protected]> > > > > wrote: > > > > > Thanks for replying. > > > > > > But how come I'm able to use a softphone to > place > > calls from withing > > > the lan? I really dont get it. What ports should > I > > enable in the > > > INPUT chain? > > > > > > > > > > > > --- On Wed, 11/18/09, Jared Smith <[email protected]> > > wrote: > > > > > >> From: Jared Smith <[email protected]> > > >> Subject: Re: [asterisk-users] can't call > through > > voip provider > > >> To: "Asterisk Users Mailing List - > Non-Commercial > > Discussion" <[email protected] > > > > >> > > > >> Date: Wednesday, November 18, 2009, 9:28 AM > > >> On Wed, 2009-11-18 at 06:01 -0800, > > >> Landy Landy wrote: > > >>> Please help me with this, I can find any > > solution on > > >> this pls help. Your help will be very > appreciated. > > Thanks. > > >> > > >> It appears that Asterisk keeps sending an > SIP > > INVITE > > >> message to your > > >> provider, but not getting any kind of > > response. After > > >> a number of > > >> attempts at re-transmitting the message, > it's > > giving up. > > >> > > >> You need to check your network configuration > and > > find out > > >> why responses > > >> from the provider aren't getting back to > your > > Asterisk > > >> system. This is > > >> typically a problem with firewalls, either on > the > > Asterisk > > >> system itself > > >> or between Asterisk and your VoIP provider. > > >> > > >> > > >> > > >> -- > > >> Jared Smith > > >> Training Manager > > >> Digium, Inc. > > >> > > >> > > >> > _______________________________________________ > > >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > >> > > >> 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 -- > > > > > > 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 -- > > > > 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 -- > > 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 -- > > 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
