Finally I got it working by removing the pfsense firewall. Something to do with pfsense firewall.
Regards On Mon, May 28, 2012 at 2:36 PM, Gopalakrishnan N < [email protected]> wrote: > Actually I understood that register line is not required, also since my > PBX is behind the pfsense firewall, now what i am going to do is putting > the PBX directly in public network (i.e. without firewall) and will check > whats going to happen. > > Hope things would sort out. > > Regards. > > > On Sat, May 26, 2012 at 2:48 AM, Stephen J Alexander <[email protected] > > wrote: > >> If your server says it is registered, that could be part of the problem. >> Vitelity doesn't use trunk registration, only IP authentication. You should >> not be using a registration string in your trunk definition. I don't know >> if it will hurt but it won't help. >> >> It sounds like you might have only 1 trunk defined, but you need 2; one >> for inbound and one for outbound. Their servers for incoming calls and for >> outgoing calls are separate. If fixing that doesn't do the job, make sure >> that incoming traffic from Vitelity is correctly routed to your PBX (and >> that they have the correct IP to send SIP traffic to). >> >> Regards, >> >> Stephen J Alexander >> MPBX, LLC >> http://mpbx.com >> 832-713-6729 >> >> >> On Fri, May 25, 2012 at 4:12 PM, Ralph Green <[email protected]> wrote: >> >>> Howdy, >>> Since the subject is Viteiy Setup, I don't think this is off topic. >>> My big problem with Vitelity is getting my server to register for >>> incoming calls. I can make outgoing calls just fine. My server says >>> it is registered with Vitelity, but no calls come in. Every attempt >>> to call the number generates an email saying there was a failed call. >>> I am using IAX, not SIP, and that is probably part of the problem. >>> IAX should work better in several ways, but few enough people use it. >>> Vitelity support has been unhelpful so far. My suspicion is that >>> there is a setting they need to make in their server so that calls go >>> to the registered IAX server, instead of looking for a SIP >>> registration, which is not there. Has anyone here worked past such a >>> problem? Was there some special thing I need to ask Vitelity? >>> Thanks, >>> Ralph >>> >>> >>> On 5/24/12, Stephen J Alexander <[email protected]> wrote: >>> > If I were troubleshooting this, the next thing I would do is verify >>> > connectivity on the relevant ports – more plainly, make sure that >>> there's >>> > not a firewall rule with unintended consequences somewhere between your >>> > asterisk and your ISP. Otherwise, as Alejandro suggests – check with >>> > Vitelity support. >>> > >>> > Regards, >>> > >>> > Stephen J Alexander >>> > MPBX, LLC >>> > http://mpbx.com >>> > 832-713-6729 >>> > >>> > >>> > On Thu, May 24, 2012 at 9:24 AM, Alejandro Imass <[email protected]> wrote: >>> > >>> >> On Thu, May 24, 2012 at 4:07 AM, Gopalakrishnan N >>> >> <[email protected]> wrote: >>> >> > yes I did that, even then i am not able to make outbound and >>> inbound as >>> >> > well. >>> >> > >>> >> > >>> >> >>> >> >>> >> That's weird. Guess you're gonna have to place a detailed ticket to >>> >> them. It sounds like a network problem to me but without any detailed >>> >> info it's hard to say. Maybe you can try sip set debug in the console >>> >> for the IP and see if you can get an idea of what is happening at the >>> >> packet level. >>> >> >>> >> We use Vitel, Skype SIP (we recently eliminated this one), and now >>> >> Gafachi and they all seem to work per there set-up instructions right >>> >> away. >>> >> >>> >> -- >>> >> Alejandro >>> >> >>> >> -- >>> >> _____________________________________________________________________ >>> >> -- 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 >>> >> >> >> -- >> _____________________________________________________________________ >> -- 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 >> > >
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