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Unread 03-26-2008, 01:36 PM   #1
ptruman
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Default Call connection, voicemail and amusing routing :)

Hi all

Ok - here's a good question for you - bear with me

I have an SPA3102, linked to my PSTN/POTS landline.
The majority (>=95%) of my inbound calls come in over PSTN.
All outbound calls go out via Voxalot - and then to my VSP or the net accordingly.

Given that I wanted one location for Voicemail, I set my SPA3102 to forward calls on busy or no answer to my Voxalot number - which won't answer, so duly goes to Voicemail - which is what I want

FYI - the SPA3102 forwards calls after 20 secs, and Voxalot Voicemail delay is set to 5 seconds, so POTS callers only have to wait 25 secs before getting to leave a message.

However, if someone calls me on my Voxalot SIP URI, and I don't grab it in 5 seconds, it will go to Voicemail, which is annoying.

Therefore - can I setup a call connection rule to detect my SPA3102 (which is registering with Voxalot on my primary account) calling Voxalot?

i.e. ONLY forward calls from myself (i.e. the SPA3102) to Voicemail and let other calls ring?

If so - would there then be anyway to forward "those" calls back to Voicemail if they don't get picked up?

(I'm after utopia here)

EDIT : I could obviously set the connection rules to forward calls "from myself to myself" only during work hours for example, but would rather it be on permanently...

EDIT x 2 : We still don't have visible call records yet - so I can't check the CLID to confirm the URI if I'm calling myself

Last edited by ptruman; 03-26-2008 at 01:48 PM.
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Unread 03-27-2008, 01:37 AM   #2
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This is what I would think would be ideal:

-Get a VoIP account with a third party (FWD, MySipSwitch, Gizmo OR anything else you can think of that's simple/open and reliable to deliver SIP)
-Register this in your VoXalot acct setting SIP register to Yes
-Use the forwarding rules to ensure all calls coming to that provider are sent to VM automatically
-Now have your SPA send PSTN callers to the third party destination after 20sec, which in turn will land them in your VoXalot VM right away

Because you are now using Call Forward rules to send PSTN callers to VM, you can raise your VM answer time for regular calls coming over VoXalot or any other of the registered providers

Now, whether that helps or not, remains to be tested

Last edited by emoci; 03-27-2008 at 01:40 AM.
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Unread 03-27-2008, 08:28 AM   #3
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Quote:
Originally Posted by emoci View Post
This is what I would think would be ideal:

-Get a VoIP account with a third party (FWD, MySipSwitch, Gizmo OR anything else you can think of that's simple/open and reliable to deliver SIP)
-Register this in your VoXalot acct setting SIP register to Yes
-Use the forwarding rules to ensure all calls coming to that provider are sent to VM automatically
-Now have your SPA send PSTN callers to the third party destination after 20sec, which in turn will land them in your VoXalot VM right away

Because you are now using Call Forward rules to send PSTN callers to VM, you can raise your VM answer time for regular calls coming over VoXalot or any other of the registered providers

Now, whether that helps or not, remains to be tested
I may be misreading this, but the PSTN delay to VM (from the ATA) is 20 secs, and because I use Voxalot VM, there is another (currently set to 5) second delay before Vox VM picks up.

In your example, am I still using the ATA to register with Voxalot? (which I am now)

The problem is the ATA sends "any" calls to VM after a given delay (20 seconds), HOWEVER, if it fwds to Voxalot VM there is another (secondary) delay - even if I'm already using the Vox account (it's receiving a call as Line1 on the ATA - so should be engaged and therefore not delay?)

Is there a way to fwd to a Voxalot VM mailbox as a SIP address directly? That would provide a different number to call, but still keep the call on the same provider, and allow my MWI to work nicely
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Unread 03-27-2008, 12:50 PM   #4
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Although there is a delay setup in VoXalot for VM, it won't apply if there is a forwarding rule that sends all calls via a provider to VM....(FW rule takes priority)

Eg.

-Your ATA remains registered as now
-You register a Gizmo acct. in VoXalot (SIP registered)
-Setup a CFW rule for all calls from Provider: Gizmo to go to VM right away
-Have your PSTN forward calls to GIZMO SIP URI after 20 sec. (this is the only delay that should apply)
-Now adjust your VoXalot VM delay to something higher, as this will apply to all other calls into your VoXalot acct., except the ones coming in via that Gizmo acct.

Does that help/clarify......
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Unread 03-27-2008, 01:53 PM   #5
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Sort of

If I set the Gizmo (for example) account to register, I would have to pay for that additional connection?

I can see the PSTN->ATA->Gizmo->Voxalot VM route working with the 20 sec delay on the ATA - and as long as I keep the Voxalot VM delay BELOW the ATA fwd delay, in theory if someone rings my Voxalot number, it should ring my ATA as normal?

Being able to address our VM mailboxes via a SIP URI would be easier, as then we could access them direct and not have to route like this
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Unread 03-27-2008, 03:30 PM   #6
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Quote:
Originally Posted by ptruman View Post
Sort of

If I set the Gizmo (for example) account to register, I would have to pay for that additional connection?

I can see the PSTN->ATA->Gizmo->Voxalot VM route working with the 20 sec delay on the ATA - and as long as I keep the Voxalot VM delay BELOW the ATA fwd delay, in theory if someone rings my Voxalot number, it should ring my ATA as normal?

Being able to address our VM mailboxes via a SIP URI would be easier, as then we could access them direct and not have to route like this
Oh, ok....I assumed you had at least 5 Sip registrations, forgot the fact that you could have VoXLite

Curious you say this
Quote:
and as long as I keep the Voxalot VM delay BELOW the ATA fwd delay, in theory if someone rings my Voxalot number, it should ring my ATA as normal?
does this mean the forwarding on your ATA applies to incoming PSTN calls as well as incoming VoIP calls.....
If that's the case, there is one more possibility:

Get a VoXBasic acct. that's not registered anywhere and dedicated solely to VoiceMail, and have the ata forward to this adresses' SIP URI

Hence all calls PSTN or VoIP will end up in that MailBox...then simply disable VM for your premium acct altogether....
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