[Asterisk-Users] dialout redunancy.

[Matthew B Marlowe]

You're toll-free number automatically forwards to the next number if one is busy? Cool. I wasn't sure if it would do that.  I know VP reports a fast busy.  Don't know what Voiceglo reports.

What toll-free provider do you have out of curiosity?

-----Original Message-----
From: asterisk-users-admin at lists.digium.com [mailto:asterisk-users-admin at lists.digium.com] On Behalf Of John Bittner
Sent: Sunday, February 08, 2004 6:12 PM
To: asterisk-users at lists.digium.com
Subject: RE: [Asterisk-Users] dialout redunancy.

I got it working by configuring qualify in my iax.conf. I guess asterisk
didn't think the IAX provider was down until I added that line.

As for incoming I have an 800 number pointing to 2 local phone numbers. 1 on
voicepulse and 1 on voiceglo. This way if voicepulse is down it will route
the call to voiceglo. Hopefully as the voip providers get better they will
offer a forwarding feature. Vonage does.

John Bittner
Simlab.net


> -----Original Message-----
> From: asterisk-users-admin at lists.digium.com 
> [mailto:asterisk-users-admin at lists.digium.com] On Behalf Of 
> Matthew B Marlowe
> Sent: Sunday, February 08, 2004 5:45 PM
> To: asterisk-users at lists.digium.com
> Subject: RE: [Asterisk-Users] dialout redundancy.
> 
> Dialout redundancy using this method works perfect.  I've 
> been using this method for some time now.  I currently have 
> two IAX2 providers and plan to get another backup as well (In 
> addition to me getting my Digium cards tomorrow that'll be 
> another backup.)
> 
> That's great for outgoing calls, but... I'm trying to figure 
> out the best approach to use for incoming calls.
> 
> I currently have a VP phone number, it's the only incoming 
> number I have for the other voip providers I have don't offer 
> local termination (or any at all for that matter).
> 
> We have a POTS line from Verizon and we'd like to continue 
> using that phone number.  
> 
> Originally we were just going to forward that phone number to 
> VP.  But what happens if VP goes down?  I figure in that case 
> (and we'd have to get in touch with VP if they will forward 
> to another number if they're done), to then forward to 
> another voip / pots line that we have.
> 
> Is there any other approach we can use to do this?
> 
> Possibly, a service that'll offer something like:
> 
> Transfer to 1609xxxxxxx but if busy, forward to 1609xxxxxxx, 
> etc. and so on?
> 
> In addition does anyone know where I might be able to port my 
> number to that supports transferring instead of forwarding?
> 
> I currently have Verizon and they said we need a CustoFlex 
> plan which will only support 6 "forwards" so if 7 callers 
> call in, the 7th will get a busy signal.
> 
> -----Original Message-----
> From: asterisk-users-admin at lists.digium.com 
> [mailto:asterisk-users-admin at lists.digium.com] On Behalf Of 
> Brent Franks
> Sent: Sunday, February 08, 2004 3:15 PM
> To: asterisk-users at lists.digium.com
> Subject: RE: [Asterisk-Users] dialout redunancy.
> 
> You will need to set priorities for each one.
> 
> For example:
> 
> exten => _91NXXNXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
> exten => _91NXXNXXXXXX,2,Playback(pstnallbusy)
> exten =>
> _91NXXNXXXXXX,3,Dial,IAX2/[PROTECTED]@voicepulse/${EXTEN:${TRUNKMSD}}
> exten => _91NXXNXXXXXX,4,Congestion
> 
> Basically what happens here, is I try to put it out on the 
> Verizon POTS
> lines first, then if that doesn't work, I play a message 
> saying all the
> lines are busy, hold if the call is important (it's now billable), the
> user holds, and it goes to voicepulse.
> 
> You could get rid of the All Busy message if you wanted, I 
> just like to
> know that the call is going to be billed (since I have unlimited LD on
> my POTS lines).  If that fails, It plays a fast busy.
> 
> You can also do a qualify in your iax.conf and sip entries to know
> whether they are reachable before trying the call. Read up on 
> qualify to
> find out how to do it for your needs.
> 
> Brent
> 
> 
> 
> -----Original Message-----
> From: asterisk-users-admin at lists.digium.com
> [mailto:asterisk-users-admin at lists.digium.com] On Behalf Of 
> John Bittner
> Sent: Sunday, February 08, 2004 2:37 PM
> To: asterisk-users at lists.digium.com
> Subject: [Asterisk-Users] dialout redunancy.
> 
> Hi,
>  
> How do I get asterisk to use an alternate outbound provider 
> in the event
> my primary IAX provider goes down. I currently have an IAX 
> provider that
> is having issues, so I signed up with a sip provider for a backup. I
> added the sip provider info into the extensions.conf file as 
> the second
> outbound entry, but asterisk still tries to call the iax provider
> 1st and since the call is incomplete the end-user hangs up. Any ideas
> would be helpful.
>  
> Thanks
>  
> John Bittner
> Simlab.net
>  
> 
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