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Unread 12-23-2007, 08:51 AM   #1
Kenthurst35
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Default Fwd SPA3102 PSTN calls to Vox Voicemail

Hi everyone

I'm new to the Voxalot service so forgive my ignorance!

I have a problem configuring my SPA-3102 to forward unanswered/busy PSTN calls to Voxalot Voicemail.

At the moment, I have Line 1 & PSTN Line set up to the same voxalot account (different ports), on which I have 1 VSP registered (GoTalk). In Voxalot, I have a dialplan which forwards all calls to my GoTalk account.

On the 3102, the PSTN-to-VoIP gateway is enabled, with an Answer delay of 15 secs. Default dialplan is 4, and this dialplan reads (S0<:xxxxxx@au.voxalot.com>) the xxxxxx representing my Voxalot userID.

After trawling through numerous forum threads here and on Whirlpool, my understanding of the call path is as follows:

PSTN call ---> internal VoIP call to Line 1 ---> rings for 15 secs (no answer) ---> PST-to-VoIP gateway takes the call ---> dialplan 4 selected, with immediate call to my voxalot account ---> call goes to voxalot voicemail.

Trouble is this doesn't work.

Ideas anyone?

Last edited by Kenthurst35; 12-23-2007 at 09:20 AM. Reason: 3102 dialplan update
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Unread 12-24-2007, 02:07 AM   #2
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Quote:
Originally Posted by Kenthurst35 View Post
Hi everyone

I'm new to the Voxalot service so forgive my ignorance!

I have a problem configuring my SPA-3102 to forward unanswered/busy PSTN calls to Voxalot Voicemail.

At the moment, I have Line 1 & PSTN Line set up to the same voxalot account (different ports), on which I have 1 VSP registered (GoTalk). In Voxalot, I have a dialplan which forwards all calls to my GoTalk account.

On the 3102, the PSTN-to-VoIP gateway is enabled, with an Answer delay of 15 secs. Default dialplan is 4, and this dialplan reads (S0<:xxxxxx@au.voxalot.com>) the xxxxxx representing my Voxalot userID.

After trawling through numerous forum threads here and on Whirlpool, my understanding of the call path is as follows:

PSTN call ---> internal VoIP call to Line 1 ---> rings for 15 secs (no answer) ---> PST-to-VoIP gateway takes the call ---> dialplan 4 selected, with immediate call to my voxalot account ---> call goes to voxalot voicemail.

Trouble is this doesn't work.

Ideas anyone?

So what is happening instead?

I would imagine what's happening is:

You call PSTN>>Rings for 15 secs>>Call is transferred to VoXalot>> VoXalot rings for about 28 secs (because you have registered your SPA with it)>>Then VM should answer

Try doing this using a fresh/new VoXalot acct that's not used or registered for anything else,just for VM... (SO xxxxxx@au.voxalot.com is a new acct)

Knowing what's happening right now would help understand where things are going wrong
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Unread 12-24-2007, 11:32 AM   #3
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"Trouble is it doesn't work"....

I can see from the info page that the PSTN Line has placed a VoIP call to xxxxxx@au.voxalot.com, but to the PSTN caller the line just keeps on ringing and is eventually answered by Telstra voicemail.

I have used Gizmo to test a SIP call to xxxxxx@au.voxalot.com and this goes immediately to voicemail, so the address appears to be valid. This occurs despite the fact that my ATA is registered to this account.

Hope this helps.
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Unread 12-24-2007, 11:35 PM   #4
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Quote:
Originally Posted by Kenthurst35 View Post
"Trouble is it doesn't work"....

I can see from the info page that the PSTN Line has placed a VoIP call to xxxxxx@au.voxalot.com, but to the PSTN caller the line just keeps on ringing and is eventually answered by Telstra voicemail.
Hmmm.... see I don't have a SPA3102 so I am not very familiar with how it works...However I had something like this happen a while ago while trying to forward callers calling a DID number registered in my PAP2 to VoXalot for Voicemail use... (this is similar but not the same thing you are doing)

Your Setup: PSTN>>15 sec. ring>>FW to VoXalot>> VM
My setup: Call on DID registered in PAP2>>FW to VoXalot >>VM

What I noticed was that calling from landlines everything seemed to work fine,VM would answer, but Mobile callers would hear infinite ringing despite the forward....(unfortunately I wasn't able to solve this, but in my case it was a DID number so I registered it directly on VoXalot and that solved my issues, in your case you want to link the PSTN to VoIP which means you can only use the SPA for this)


Quote:

I have used Gizmo to test a SIP call to xxxxxx@au.voxalot.com and this goes immediately to voicemail, so the address appears to be valid. This occurs despite the fact that my ATA is registered to this account.

Hope this helps.
This may not be related to your problem but it bothers me that a Gizmo call does not ring your ATA but rather goes directly to VM since it means your SPA may not be registered with VoXalot properly....

I would suggest the following:

-Reset your SPA if you know how to do so without causing any problems, and provided you didn't require any special settings to set it up to start with (pretty much if there is any settings that either a provider or someone else set up for you to get it to work, you probably don't want to do a reset)

-If you can't do the reset safely, and least powercycle it, turn it off and on...

-Ensure you can make and receive calls via VoXalot and the GoTalk acct within it (do not enable the PSTN to VoIP gateway yet or have PSTN hooked up yet, ensure you can use the SPA as a normal ATA by registering it to VoXalot and try making some calls, to try receiving calls use a SipBroker Access Number and dial *010YourVoXNumber when prompted, if everything goes well your ATA should ring)

-Now enable PSTN, and hook up, and at first simply ensure that PSTN calls coming in can be received via the SPA and there is 2 way audio

-If things up to this point look good, try forwarding again, but use a VoXalot acct that's new and not used/registered anywhere else on the setup

This is pretty much a step by step trial of the SPA3102 functions in increasing complexity to see if you can find the culprit... Try forwarding to xxxxxx@voxalot.com, xxxxxx@au.voxalot.com, xxxxxx@us.voxalot.com and see if one gives better results...

Also try forwarding to your gizmo acct user@proxy01.sipphone.com (user can be your username or your 747 number) and see if it rings there when a PSTN call comes in


My overal thoughts are this:

(1) This a similar problem like I encountered with the PAP2,the Telstra line doesn't like redirections
(2) The reason it keeps ringing is that you are forwarding to VoXalot acct already registered which means VM will answer after about 28 secs by default unless you've changed it
(3) There is some sort of conflict forwarding to SPA>>VoXalot registered in SPA>>which leads back to SPA (sort of a loopback effect)
(4) There is a setting on your SPA you are overlooking

This is a slow time so not many people hang around but if some of the SPA users do drop in, they may be able to give more info and/or pointers on this...

I don't know how helpful I was....

Last edited by emoci; 12-24-2007 at 11:38 PM.
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Unread 12-25-2007, 10:42 PM   #5
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Hi emoci

Thanks for the detailed response. I really appreciate the help.

Quote:
Originally Posted by emoci View Post
This may not be related to your problem but it bothers me that a Gizmo call does not ring your ATA but rather goes directly to VM since it means your SPA may not be registered with VoXalot properly....
I've checked and actually the Gizmo call does ring my ATA, its just that I had the Voicemail delay set really low on my Voxalot account, so it "appeared" to be going straight to VM.

As regards other testing.....

I took your advice and set up another Voxalot basic account, substituting those details in the PSTN tab. Same result thought, the PSTN line just rings out after the gateway picks up the call. VM on the new account never answers. I've checked, and I have IP Dialling set to YES on the Line 1 tab and the manual says that this will apply to the PSTN Line as well, but I will test to see whether in fact I do have a problem with IP dialing generally.

Haven't reset the device as yet coz its a bit of a pain entering all the ozzie settings, but that will be what I try next if I can't fix this thing.

However, i'm intrigued by your comments about not being able to get calls from mobiles to divert. I've been using my mobile for the PSTN to VM testing, so what might the difference be? Anyhow, I will find another landline to ring the PSTN and see if that makes a difference.

Cheers.
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Unread 12-27-2007, 11:15 AM   #6
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OK, I've done some more testing.....

With a new Voxalot account specifed in the PSTN Line dialplan, as I've said before the line appears to the PSTN caller to just ring out. Curiously though, after each call I am getting an email from Voxalot as below:

"Hello:

just wanted to let you know you were just left a 0:04 long message (number 5) in mailbox xxxxxx from Unknown Caller, on Wednesday, December 26, 2007 at 04:12:54 AM, so you might want to check it when you get a chance by dialing extn *500. Thanks!

--Voxalot Voicemail Server"

Each time a run this test, I get a new message identical to the one above, including the supposed message duration of 0:04. Strange huh?

I have also tried specifying my sipphone account on the PSTN tab, and I find that when I ring through on PSTN it does ring the Gizmo app running on my PC. Of course this may not be a valid test since both the ATA and PC running Gizmo are on the same LAN.

What I am beginning to wonder is if this problem is due to the fact that I am using IP dialling, but have no STUN server setup and NAT Mapping Enable set to NO. I have read a number of threads detailing problems of 1-way audio/loss of RTP packets and I am wondering if I have this problem since according to Voxalot I am leaving voicemails on the new account I set up.

Presumably, NAT/STUN server settings are not as important if the ATA is doing a SIP registration, and that is why I up until now I have not been having problems making or receiving VoIP calls using the ATA.

Am I getting anywhere here?
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Unread 12-28-2007, 03:30 AM   #7
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Quote:
Originally Posted by Kenthurst35 View Post

What I am beginning to wonder is if this problem is due to the fact that I am using IP dialling, but have no STUN server setup and NAT Mapping Enable set to NO. I have read a number of threads detailing problems of 1-way audio/loss of RTP packets and I am wondering if I have this problem since according to Voxalot I am leaving voicemails on the new account I set up.

Presumably, NAT/STUN server settings are not as important if the ATA is doing a SIP registration, and that is why I up until now I have not been having problems making or receiving VoIP calls using the ATA.

Am I getting anywhere here?
Well at least your SPA is indeed forwarding....

I don't know if the STUN, NAT Mapping enable will help in this case... I know they won't hurt since I've got both set up in my PAP2 without issues

Might as well give it a try.....
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Unread 12-28-2007, 11:49 AM   #8
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Tried using STUN server settings (used Voxalot server), no difference.

Same behaviour from the new Voxalot account I set up. To the PSTN caller the line just rings out, but from Voxalot's perspective I have activated the voicemail on that account, hence the emails.

I am really beginning to suspect that the issue lies with my ATA not detecting a call has been answered (and/or no audio) when that call is made via IP dialling. If it is that specific, surely I am not the only person to have experienced this?

Come on guys, there have been over a 100 viewings of this thread. Someone out there must know something!

Cheers
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Unread 12-28-2007, 12:37 PM   #9
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its a bit long thread and im not so sure what stage you guys already at, but as i understand is, all you want is to forward a busy and/or unanswered pstn calls to vox vm, if thats the case you should input the vox number in user 1 tab.
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Unread 12-28-2007, 11:05 PM   #10
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Hi dc2007, thanks for the reply.

Quote:
Originally Posted by dc2007 View Post
its a bit long thread and im not so sure what stage you guys already at, but as i understand is, all you want is to forward a busy and/or unanswered pstn calls to vox vm, if thats the case you should input the vox number in user 1 tab.
I think this is the call routing logic you are describing above...

PSTN call ---> internal VoIP call to Line 1 ---> busy/no answer ---> call forward to destination on User 1 Tab (entered as <voxalot number>@au.voxalot.com:<port>) ---> call goes to voxalot voicemail.

Yeah, I tried this, but all that happened is the PSTN gateway eventually cut in and gave me dial tone. Obviously i had set the CFwd No Answer delay to be much shorter than the gateway delay.

Can anyone confirm what happens if you try to CFwd to the Voxalot account registered on Line 1? Some threads I have read suggest that it will go straight to VM, whereas others say that it just creates a kind of look back to the ATA.

Cheers
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