Greg,
> At some point you need to consider how much is too much...
I agree. Up until a few days ago, we thought (and were told by our
"major" provider) that their network is extremely reliable. Over the
weekend all the T1s to our stores were down. Our bonded T1s were supposed to be
redundant - they went down together. That was fixed after many hours, and the
next day they had an unrelated major outage which took out all our DIDs (T1s
were up, but no calls coming thru from the provider).
We are in the process of getting fiber from another provider like you for at
least some critical locations - that will alleviate the connectivity issues, but
our biggest concern is the DIDs now.
Matt
> Date: Tue, 30 Apr 2013 23:33:22 -0500
> From: gmalsack at coastalacq.com
> To: asterisk-users at lists.digium.com
> Subject: Re: [asterisk-users] multiple provider for incoming
>
> Matt,
>
> At some point you need to consider how much is too much...
>
> I run a call center with more then 125 commissioned phone sales reps and
more than 60 customer service reps. We run dual servers, fiber from one provider
and 6 bonded T1's from another provider. We purchase our so trunks from a
wholesale company who is a major provider to resellers. Being so, their network
is extremely reliable. However, late last week an upstream/downstream provider
had am outage which affected some of our DIDs but not others.
>
> Your assumption that porting a number from one provider to another is
correct. If I remember correctly it's an FCC mandate that a number cannot be
ported within 30 days of a previous port.
>
> Greg
>
> Matt Hamilton <mistral9999 at hotmail.com> wrote:
>
> >Don,
> >
> >Inbound reliability is very important. We don't use toll-free
numbers, but we will look into that. I thought porting numbers - not sure about
toll-free though - from one provider to the other took days (not technically,
but paperwork, etc.)
> >
> >Thanks,
> >Matt
> >
> >From: dk at donkelly.biz
> >To: asterisk-users at lists.digium.com
> >Date: Tue, 30 Apr 2013 22:38:44 -0500
> >Subject: Re: [asterisk-users] multiple provider for incoming
> >
> >If inbound reliability is important, you may be able to accomplish what
you want with redundant servers, multiple sip providers and toll-free numbers
that can be readily switched between the sip providers.--Don From:
asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at
lists.digium.com] On Behalf Of Matt Hamilton
> >Sent: Tuesday, April 30, 2013 10:25 PM
> >To: Asterisk Users Mailing List - Non-Commercial Discussion
> >Subject: Re: [asterisk-users] multiple provider for incoming >The
process will depend on your provider, of course, but I know some have an option
that if they are unable to reach
> >>your box, then they can auto-forward to another DID, or to a
voicemail box, or to a user-defined function, etc.
> >
> >Forwarding to another DID will/should work for us assuming they are
going to be able to do that during a failure on their side. During a recent
outage (I think they had some major issues at one of their switches), they were
not able to send the calls to our box which was online.
> >
> >Thanks,
> >Matt
> >
> >Date: Tue, 30 Apr 2013 20:38:19 -0500
> >From: wcselby at selbytech.com
> >To: asterisk-users at lists.digium.com
> >Subject: Re: [asterisk-users] multiple provider for incomingOn Tue, Apr
30, 2013 at 7:50 PM, David Wessell <david at ringfree.biz> wrote:Hi Matt,
You can't have multiple providers for inbound traffic. You can have multiple
providers for outbound traffic though. ThanksDavid David, I'm not sure
where you got this information, but it's not accurate. I've had
multiple inbound and outbound SIP providers for years going to a single box.
You just get a separate DID from each provider. Matt,
> >
> >The process will depend on your provider, of course, but I know some
have an option that if they are unable to reach your box, then they can
auto-forward to another DID, or to a voicemail box, or to a user-defined
function, etc.
> >--
> >Thanks,
> >--Warren Selby, dCAP
> >http://www.SelbyTech.com
> >--
_____________________________________________________________________ --
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
> --
> _____________________________________________________________________
> -- 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/20130501/66a71bc9/attachment.htm>