Voxalot and Sipura/ATA Tutorial: A Comprehensive Walkthrough

[voip2007]

This configues works good for me. But I still have a problem annoying me. I think it is the issue of firewall/NAT. I miss 30% incoming call which didn't ring to my phone or go directly to my voicemail. How should I change the configue?

[martin]

Originally Posted by voip2007

This configues works good for me. But I still have a problem annoying me. I think it is the issue of firewall/NAT. I miss 30% incoming call which didn't ring to my phone or go directly to my voicemail. How should I change the configue?

What is the registration expiry setting on your phone / device?
.

[voip2007]

I set the registration expire as 600. and I also tried to set 60. the results are the same.

[BigTex]

Many, many thanks for this excellent tutorial! For Vitelity inbound service, I found that I needed to make a change. For my Vitelity account, your instructions led me to conclude that I should use inbound6.vitelity.net as the host. But it didn't work - it never would register. However, when I used sip6.vitelity.net (which is the host name that Vitelity led me to use for connecting my ATA directly), it worked.

[jerm]

I am confused by the UserID settings.
In the Sipura-941 manual it says :

Display Name : Subscriber’s display name to appear in caller-id
User ID : Subscriber’s user-id. Usually a E.164 number
Password : Subscriber’s a/c password
Auth ID : Subscriber’s authentication ID
Use Auth ID : If set to “yes”, the pair <Auth ID> and <Password> are
used for SIP authentication. Else the pair <User ID> and
<Password> are used.

I interpret this as meaning, I'd put my name in Display Name, my DID in User ID (both for Caller ID). My login ID in Auth ID and set Use Auth ID to 'yes'.

However, I find that regardless of the setting for Use Auth ID, I always must have my login ID in User ID otherwise registration fails.

This means that my Caller ID ends up being my login ID, which is not useful for working out how to call me back, specially calling across networks.

Is this my misinterpretation, or a bug somewhere ?

Thanks for any clarification.

[jerm]

I am trying to amalgamate all voicemail into one mailbox.
For Voip Calls, using voxalot, this is relatively easy ....

For PSTN, it is not quite so ...... here is the setup.

An SPA3000 ATA on the PSTN line, whose dialplan does an automatic 'hotline' style call to the IP address of an SPA941 IP phone's Extension 1, without picking up the line. So far so good, when calls come in on the landline, the IP phone rings. (NB. I use this technique, so that these calls do not have to traverse the net via my VSP).

On this phone, Ext 1, is the only one that can do CallForward, I have Ext 1 registered to voxalot. I set up a duration in Cfwd No Ans Delay and my voxalot ID in Cfwd No Ans Dest.

The problem I face now, is that the SPA941 refuses to call through to my voicemail using my voxalot ID, because it is registered using that ID. In fact it refuses to call that number if any of the other extensions is registered using that ID.

Is there some special dial-code for a registered phone to forward immediately to the account's voicemail box?

Is there some trick using voxalot dial-plans to make the SPA941 think it is calling a different number?

Thanks for any suggestions.

[emoci]

Originally Posted by jerm

I am trying to amalgamate all voicemail into one mailbox.
For Voip Calls, using voxalot, this is relatively easy ....

For PSTN, it is not quite so ...... here is the setup.

An SPA3000 ATA on the PSTN line, whose dialplan does an automatic 'hotline' style call to the IP address of an SPA941 IP phone's Extension 1, without picking up the line. So far so good, when calls come in on the landline, the IP phone rings. (NB. I use this technique, so that these calls do not have to traverse the net via my VSP).

On this phone, Ext 1, is the only one that can do CallForward, I have Ext 1 registered to voxalot. I set up a duration in Cfwd No Ans Delay and my voxalot ID in Cfwd No Ans Dest.

The problem I face now, is that the SPA941 refuses to call through to my voicemail using my voxalot ID, because it is registered using that ID. In fact it refuses to call that number if any of the other extensions is registered using that ID.

Is there some special dial-code for a registered phone to forward immediately to the account's voicemail box?

Is there some trick using voxalot dial-plans to make the SPA941 think it is calling a different number?

Thanks for any suggestions.

Not sure whether you'll end up going to VM as you wish, but this may at least fix the problem with the ATA refusing to place the call:
Try entering 123456@us.voxalot.com or *010123456@sipbroker.com as the no answer destination (123456 stading for your VoXalot ID)

A few other thoughts:
Activate DND (Do Not Disturb feature),however this may affect all lines on that ATA, or lower the time to forward to VM (within your VoXalot acct) to just a few seconds....

What I am unclear about is how this works for you when you actually want to answer the calls from PSTN, wont this send them to VoiceMail directly?

[emoci]

Originally Posted by jerm

I am confused by the UserID settings.
In the Sipura-941 manual it says :

Display Name : Subscriber’s display name to appear in caller-id
User ID : Subscriber’s user-id. Usually a E.164 number
Password : Subscriber’s a/c password
Auth ID : Subscriber’s authentication ID
Use Auth ID : If set to “yes”, the pair <Auth ID> and <Password> are
used for SIP authentication. Else the pair <User ID> and
<Password> are used.

I interpret this as meaning, I'd put my name in Display Name, my DID in User ID (both for Caller ID). My login ID in Auth ID and set Use Auth ID to 'yes'.

However, I find that regardless of the setting for Use Auth ID, I always must have my login ID in User ID otherwise registration fails.

This means that my Caller ID ends up being my login ID, which is not useful for working out how to call me back, specially calling across networks.

Is this my misinterpretation, or a bug somewhere ?

Thanks for any clarification.

VoXalot does not use Auth (Auth can be set to No) , and you'll find that most voip providers use the User ID as the login authentication feature (in most if not all cases User ID and Auth ID are one and the same value)

Based on your provider, the CID number sent is the one registered with that account.

If you are using VoXalot with third party providers, and they support setting your own CID Number (most don't), there is a From User field in advanced settings that will let you specify a DID number....

I don't know if that helps....

[jerm]

Originally Posted by emoci

VoXalot does not use Auth (Auth can be set to No) , and you'll find that most voip providers use the User ID as the login authentication feature (in most if not all cases User ID and Auth ID are one and the same value)

Based on your provider, the CID number sent is the one registered with that account.

If you are using VoXalot with third party providers, and they support setting your own CID Number (most don't), there is a From User field in advanced settings that will let you specify a DID number....

I don't know if that helps....

Many thanks for your response.
This is strange, I would have thought that those settings on the ATA were to control how the ATA worked, not Voxalot.

Thanks for the tip about setting the CID from Voxalot, I had not noticed that feature.

[jerm]

Originally Posted by emoci

Not sure whether you'll end up going to VM as you wish, but this may at least fix the problem with the ATA refusing to place the call:
Try entering 123456@us.voxalot.com or *010123456@sipbroker.com as the no answer destination (123456 stading for your VoXalot ID)

A few other thoughts:
Activate DND (Do Not Disturb feature),however this may affect all lines on that ATA, or lower the time to forward to VM (within your VoXalot acct) to just a few seconds....

What I am unclear about is how this works for you when you actually want to answer the calls from PSTN, wont this send them to VoiceMail directly?

Thanks, but the first suggestion does not work as the Phone expects a phone number, not a sip address, it just strips out all non-digits, also unfortunately during CFWD it uses no dialplans.

I suspect that activating DND on the ATA, will stop it from doing the 'hotline' call to the IP phone.

To keep the whole thing inside my own network, the ATA is not registered to a VSP and only the PSTN setup is used. It sends the call directly to the IP address of my IP phone, without lifting the hook, by using a dialplan : <S0:nnnnnn@xxx.xxx.x.xx> where nnnnnn is the value of the User-ID of Extension 1 on the IP phone (and xxx.xxx.x.xx is it's IP address). This all works without a hitch.
The problem is getting this IP call out to VM if I do not answer. This is where I cannot get the IP phone to CFWD to the VM box on the extension which is registered using the same account.

Maybe that is clearer (crikey this stuff is complicated)