[asterisk-users] Vitelity Setup

Gopalakrishnan N gopalakrishnan.an at gmail.com
Mon May 28 04:06:57 CDT 2012


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
<sjalexander at mpbx.com>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 <sirable at gmail.com> 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 <sjalexander at mpbx.com> 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 <ait at p2ee.org> wrote:
>> >
>> >> On Thu, May 24, 2012 at 4:07 AM, Gopalakrishnan N
>> >> <gopalakrishnan.an at gmail.com> 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
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120528/83bd3a07/attachment.htm>


More information about the asterisk-users mailing list