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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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Unread 03-27-2008, 04:03 PM   #7
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Quote:
Originally Posted by emoci View Post
Oh, ok....I assumed you had at least 5 Sip registrations, forgot the fact that you could have VoXLite
Yes indeed - I (so far) only need 1 VSP.

Quote:
Originally Posted by emoci View Post
does this mean the forwarding on your ATA applies to incoming PSTN calls as well as incoming VoIP calls.....
Technically, ANYTHING ringing on the ATA will get fwd'd after the delay - however, because my ATA is registered with Voxalot, if it fwd's to itself, so, to not keep them waiting, I lower the VM delay, so they only get 20secs + 5 secs VM delay - then drop to VM.

This is fine for PSTN or VSP calls, but if someone rang me from Voxalot, then I only get 5 secs to answer the call.

Quote:
Originally Posted by emoci View Post
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....
I'd almost arrived at this conclusion also - but have another following two questions :

1) I agree I can fwd all calls to a secondary Vox basic account - but if my ATA is regged to my premium one, and I dial *500 - it's going to connect to my premium VM account.

Could I solve this by setting up the 2nd VoxBasic account as a "provider", and directing *500 to "it"?

2) My MWI setting on my ATA is set to use my premium account (which is also the one the ATA registers to)

If I change VM to another account, but leave my ATA subbed to the premium one, will I lose my MWI functionality?


Agh, I'm going to have to diagram this for myself, it's getting amusing...
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Unread 03-27-2008, 05:19 PM   #8
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Quote:
Originally Posted by ptruman View Post
Yes indeed - I (so far) only need 1 VSP.



Technically, ANYTHING ringing on the ATA will get fwd'd after the delay - however, because my ATA is registered with Voxalot, if it fwd's to itself, so, to not keep them waiting, I lower the VM delay, so they only get 20secs + 5 secs VM delay - then drop to VM.

This is fine for PSTN or VSP calls, but if someone rang me from Voxalot, then I only get 5 secs to answer the call.



I'd almost arrived at this conclusion also - but have another following two questions :

1) I agree I can fwd all calls to a secondary Vox basic account - but if my ATA is regged to my premium one, and I dial *500 - it's going to connect to my premium VM account.

Could I solve this by setting up the 2nd VoxBasic account as a "provider", and directing *500 to "it"?

2) My MWI setting on my ATA is set to use my premium account (which is also the one the ATA registers to)

If I change VM to another account, but leave my ATA subbed to the premium one, will I lose my MWI functionality?


Agh, I'm going to have to diagram this for myself, it's getting amusing...
Yeah, unfortunatelly if you do this you'll lose your MWI setting...

As for the *500, try using *010*500 which should then ask you for VoXalot number and password....better yet create a Speed Dial that'll call *010*500 via SipBroker (not sure if this will get around limited access only to one acct., will test when I get back today)


One thing though, isn't there a PSTN to VoIP Bridge function that doesn't require you to use forwarding on the SPA at all

What I mean is have it so PSTN always rings the VoIP line, which should ring your ATA...then goes to VM as any other call to VoXalot would


Edit:
I was re-reading your first post, your call form self to self idea is not bad...

Set up two Call Forwarding rules

1. Selecting on CID _123456 (where 123456 is your VoXalot number) and it sends all these calls to VM automatically
2. Create a second rule for Any Calls to 'Ring Your Account' so that you ensure all other calls ring VoXalot then get sent to VM after xx seconds

Then set up your VM delay in voxalot to 25-30, and test a PSTN call....if all works well when calling PSTN number it'll ring for 20 sec and then go to VM directly. For calls coming in over VoXalot they'll ring for however long your VM delay is then go to Vm as usual

Last edited by emoci; 03-27-2008 at 05:30 PM.
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Unread 03-27-2008, 10:52 PM   #9
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Quote:
Originally Posted by emoci View Post
Yeah, unfortunatelly if you do this you'll lose your MWI setting...
Bum!

Quote:
Originally Posted by emoci View Post
Edit:
I was re-reading your first post, your call form self to self idea is not bad...

Set up two Call Forwarding rules

1. Selecting on CID _123456 (where 123456 is your VoXalot number) and it sends all these calls to VM automatically
2. Create a second rule for Any Calls to 'Ring Your Account' so that you ensure all other calls ring VoXalot then get sent to VM after xx seconds

Then set up your VM delay in voxalot to 25-30, and test a PSTN call....if all works well when calling PSTN number it'll ring for 20 sec and then go to VM directly. For calls coming in over VoXalot they'll ring for however long your VM delay is then go to Vm as usual
I *think* I'm following, however, I've made a slight mod

a) I've left my ATA to forward to my Voxalot number (that it's registered with)
b) My ATA PSTN gateway is registered using my Voxalot number +1 (i.e. primary FX0 = 12345 and my PSTN FXS is 12346)
c) I've changed the _12345 rule you suggested to _12346 and it's working for PSTN.

I'll try inbound from SIP in a second
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Unread 03-28-2008, 09:38 AM   #10
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[EDITED FOR CLARITY]

Hmm, well it's 75% working.

My ATA "Line1" is registered with my Voxalot premium account.
My ATA "PSTN" line is registered with a Voxbasic account.

Busy/unanswered incoming PSTN calls are forwarded (by my ATA) using the Voxbasic account - to my VoxPremium number.

I've got a call rule set (as per your suggestion) on my VoxPremium account to catch calls from the VoxBasic CLID and fwd direct to VM - that IS working for inbound PSTN calls.

Incoming calls direct to my VoxPremium SIP URI are also ringing the handsets, and fwding to VM correctly (I do need to check what happens if both fwd/answers timers are the same - the ATA delay is 25 decs, the VoxPremium VM delay is 20 secs)

Whats NOT working is the rule for "everything else" to ring my VoxPremium account ("Your Voxalot Account")

Example : I have a VSP who provides me with a PSTN DID, which connects to the VSP, which is registered via my VoxPremium account, so connects back to my ATA and my handsets ring.

However, after 20/25 secs, the call drops.

Is this because incoming net calls (which reach the ATA) are trying to be forwarded (by the ATA) at 20 secs, to my own VoxPremium number, causing a call/routing loop and failing?

FYI as mentioned, my VoxPremium and VoxBasic accounts are 1 digit different, i.e. 123456 and 123457

So (bear with me again) :

a) I've edited the call rule to check for my VoxBasic CLID (which is handling PSTN calls) to just check for numbers beginning 12345
b) I think the VSP passing the originating CLID stops that rule working, so it's hitting rule 2 (all calls/providers), and passing them to my VoxPremium account

So - either one of these two things is happening :

c) The ATA rings - waits 25 seconds and tries to fwd to my VoxPremium number (which it's already connected by) - call drops.
OR
d) Even though my VSP is connecting to me VIA my VoxPremium number, the 20 sec VM delay isn't cutting in for some reason - but in that case, I'd expect a permanent ring - so I'm looking at (c)...

I have one idea - would this work :

If I used the ATA and VoxPremium call fwd rules to fwd my calls to the VoxBasic account, which IS registered (as the PSTN line) on the ATA, would MWI for that still work, or does SPA3102 MWI only work on the Line1 account?
In any case thats still not ideal as then I don't get web access to VM as it's a basic account.

There has to be a way to do this

Thoughts?

Need logs!

Last edited by ptruman; 03-28-2008 at 10:50 AM.
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