[asterisk-users] Vitelity Setup
Danny Nicholas
danny at debsinc.com
Fri May 25 16:18:04 CDT 2012
Is your IAX2 peer registered?
-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Ralph Green
Sent: Friday, May 25, 2012 4:12 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Vitelity Setup
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
More information about the asterisk-users
mailing list