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Unread 09-22-2006, 02:23 AM   #1
usf
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Default sip register select "yes" ? while add provider

After login voxalot, go to add voip provider to provider menu.
In the last step, there is a choice " yes/no " in sip register. Its default is "no".

In what condition, I should select "yes" ? there is no explain in tution.
Any member knows that.
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Unread 09-22-2006, 02:52 AM   #2
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Hello,

You only need to set this field to "yes" if you want to receive inbound calls via this provider or if the provider requires you to be registered to make calls.

At the moment changing this field can take up to 24 hours to take effect. The good news is our new registration function (to be released very soon) is looking good and registrations will soon be back to 5 mins.
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Unread 09-22-2006, 02:55 AM   #3
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If your VoIP provider supports incoming calls, you will need to set SIP Register = Yes.

Some VoIP providers require a sip registration to make outgoing calls. If yours does, you will need to set SIP Register = Yes.

Changes to this setting may take up to 24 hours to occur.

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Unread 09-22-2006, 06:17 AM   #4
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helping reply.

Martin, Ron
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Unread 09-23-2006, 10:48 AM   #5
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I register stanaphone with sip register with yes. after a few hour later, I check its status and it shows yes.
I set up voxalot in ATA. ATA can receive stanaphone ring tone . But I pick up the phone's handset, I cannot talk and receive voice.

also , I register with freedigits with sip register with yes, but sip register does not work.
freedigts need stun or outbound proxy setting. I do not see voxalot provide this type of setting. Does it cause freedigit not working ?
any member add stanaphone and freedigits with sip register "yes" in provider menu before and what is your result, work or not?
please comments.
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Unread 09-23-2006, 11:46 AM   #6
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I use both Stanaphone and FreeDigits through VoXaLot and seldom have any problems.

Your audio problems with Stanaphone are likely related to NAT issues with your router. You can confirm this by temporarily connecting your ATA directly to your Cable/DSL modem. If the problem goes away, you will know it's your router. I have to forward the appropriate SIP and RTP ports to my PAP2 as well as use a STUN server for 100% reliable audio.

FreeDigits does not need an outbound proxy server. I use proxy.freedigits.net as the Host in my Provider entry.

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Unread 09-24-2006, 12:44 AM   #7
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Thanks Ron.
I have more question.
for use in inbound call only, besides add my Voip provider in provider menu.
Need I add any more setting?
e.g Need I add any dial plan in dial plan menu or not ? or
need I press any button in phone set before I pick up the handset to talk?
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Unread 09-24-2006, 01:01 AM   #8
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Inbound calls should require nothing more than a successful SIP registration on the Providers page.

Dial Plans are used for outbound calling only.

An incoming call should ring the telephone in the normal manner and simply picking up the receiver should establish the call.

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Unread 09-24-2006, 01:15 AM   #9
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Ron ,
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need I set up stun server setting in ATA for voxalot.com for inbound call?
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Unread 09-24-2006, 02:37 AM   #10
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Quote:
Originally Posted by usf
need I set up stun server setting in ATA for voxalot.com for inbound call?
It generally depends on the particular router and ATA. Some people get away without any ports forwarded in their router and without the need to use a STUN server. Others have audio problems unless they use one or the other. In my case, I seem to need both to keep Stanaphone happy.

It's probably best to sneak up on it. See how you do without any special measures. If it works fine, enjoy it. If you have audio problems, try bypassing your router temporarily to see if it's a router issue. If it is, see if a STUN server will correct the problem. If it doesn't, try forwarding the appropriate SIP/RTP ports in your router.

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