VOIP questions

No wonder it's not working at 60 minute intervals. Mine is set to 30 seconds.
Thanks.

I've set Register Expiration: to 1 which is in minutes and is the lowest value I could use (unless I can use 0.5? Edit: can't 1 is lowest). The options for it are: (in minutes. default 1 hour, max 45 days).

If this is such a critical thing why is it set so high to begin with? Also is there a way to find out how quickly it stops working? If for instance it's every 45 seconds that means there will be 15 seconds per minute people can't phone in.

Ok That hasn't worked. I can call in just after updating but after a minute or so it fails again.

I have also found SIP OPTIONS/NOTIFY Keep Alive Interval: which is set to 30 seconds. I will have a play with lowering this.

EDIT: which I lowered to 10 and still have the "going to sleep" issue.
 
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Ok. I had it working. Called the number and phone rang. I had a play with call recording setting in A&A control panel and now the phone doesn't ring anymore. I've turned it all off but still doesn't ring. FFS.

Hmm.. now it's working again.

Ok what I seem to be getting is that calling in works for a short time and then stops working. If I call out I can then call in again for a short time and then it stops working again. It's like it's going to sleep.
Can you decrease the registration timer. Had this previously on SIP trunks, where it would work for incoming when it had just registered as it had punched a hole in the firewall for its incoming comms, only problem the firewall was set to close those opened ports after 2 minutes and the SIP only registered every 5, so for 2 minutes it worked, then 3 it didnt, then 2 it did and so on. we change the registration to 2 minutes and no problems after that.
 
Thanks for all the help.

I've changed Register Expiration: to 1 minute and SIP OPTIONS/NOTIFY Keep Alive Interval: to 1 second. Both are the lowest value aloud. Just after I update I can phone in. A few seconds later and it fails.

This is driving me nuts.
 
I wouldn't be surprised if your mobile network is very aggressively closing NAT sessions and you can't make this work
Is this likely to be the same for different networks? I'm testing it on lebara but I intend to deploy it on three.

I still have standard broadband at work alongside the mr600. My plan was to to get this up and running on the mr600 and then port the number and close the standard broadband down.

I will take it to work tomorrow and test it on both routers. If it works on the normal bb but not the 4g+ I will have to have a rethink.
 
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Thanks for all the help.

I've changed Register Expiration: to 1 minute and SIP OPTIONS/NOTIFY Keep Alive Interval: to 1 second. Both are the lowest value aloud. Just after I update I can phone in. A few seconds later and it fails.

This is driving me nuts.

I hate to say it but VOIP drove me mad.

You wait until you phone your bank and they lock all your accounts because you are using a VOIP phone. I mean, seriously, the UK is not ready for VOIP yet.
 
VoIP itself is fine if done correctly and not using SIP on non static IP or even non dedicated IP connections.

I work for a Communications company and we have over 100 customers using VoIP telephone systems all over Europe and North America. We use a single supplier who don't rely on SIP and so the apps on phones work on all internet connections using push and as long as you don't block outbound traffic on your router/firewall the physical handsets work without issues as well.
 
I hate to say it but VOIP drove me mad.

You wait until you phone your bank and they lock all your accounts because you are using a VOIP phone. I mean, seriously, the UK is not ready for VOIP yet.

Your bank would have no way of telling that you were using a VoIP phone. The majority of business calls have been VoIP for a decade, the core telephone network has been IP for about that long as well.
 
Update time:

I took it all to work and tried it on the TalkTalk router and it worked fine (I had to disable ALG SIP). I then tried it on the MR600 with the three sim and again it was fine. So it appears to be a lebara issue. It's good job I didn't spend several hours trying to get it to work at home, oh wait a minute...

Thanks again for all the help.
 
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I want to purchase a Grandstream HT802.
have you(& your customers) been happy with it's performance, specifically the quality of the audio it provides via it's analogue/traditional phone plug ? or has using a voip app on your mobile provided better quality & HD sound.
BT provide a similar adaptor with their hub and people criticize quality, and demand a dect phone that will work directly with their hub.

A&A call rates seem higher than localphone I currently use for outgoing calls 1.5p/4p vs 0.6/1.8p
(https://www.aa.net.uk/voice-and-mobile/voip-information/, https://www.localphone.com/prices/u#pricesTableContainer)



VoIP itself is fine if done correctly and not using SIP on non static IP or even non dedicated IP connections.
probably a naive Q - Should you obtain a static IP on talk talk say and would that ensure a more reliable connection for incoming calls
 
have you(& your customers) been happy with it's performance, specifically the quality of the audio it provides via it's analogue/traditional phone plug ? or has using a voip app on your mobile provided better quality & HD sound.
BT provide a similar adaptor with their hub and people criticize quality, and demand a dect phone that will work directly with their hub.

A&A call rates seem higher than localphone I currently use for outgoing calls 1.5p/4p vs 0.6/1.8p
(https://www.aa.net.uk/voice-and-mobile/voip-information/, https://www.localphone.com/prices/u#pricesTableContainer)
I haven't used it with the business number as I'm currently waiting for it to port over, but I've been very happy with the test account. I set up groundwire on my mobile and it appears to work very well. Tbf it couldn't be any worse than our talk talk line. I'm sure there must be a fault with it, it's that bad, but they say not.

I didn't really look at local phone. I will keep it in mind in case I don't get on with A&A. Edit: their website is terrible on mobile.
 
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have you(& your customers) been happy with it's performance, specifically the quality of the audio it provides via it's analogue/traditional phone plug ? or has using a voip app on your mobile provided better quality & HD sound.
BT provide a similar adaptor with their hub and people criticize quality, and demand a dect phone that will work directly with their hub.

I've got an HT-801 (same thing, just the 1 port version) and it's absolutely fine. Call quality is very good. Not that it gets much use these days.

probably a naive Q - Should you obtain a static IP on talk talk say and would that ensure a more reliable connection for incoming calls

Not necessary. My WAN IP is surprisingly dynamic and over years of using VoIP (with Gradwell & Sipgate) I've had zero issues that were down to the dynamic IP.
 
A decent VoIP provider today will be using username/password for you to login and then hopefully push for incoming calls so the public IP address or location of the user shouldn't matter. I have an 8x8 VoIP service (its business, but I use it at home and on my mobile) and never have to worry about whether ill receive incoming calls or not.

This is generally different to standard SIP that uses ports and forwarding and is reliant on a stable public IP address and firewall setup.
 
A decent VoIP provider today will be using username/password for you to login and then hopefully push for incoming calls so the public IP address or location of the user shouldn't matter. I have an 8x8 VoIP service (its business, but I use it at home and on my mobile) and never have to worry about whether ill receive incoming calls or not.

This is generally different to standard SIP that uses ports and forwarding and is reliant on a stable public IP address and firewall setup.

SIP should be fine too. The Gradwell and Sipgate services that I've been using for 12 years or so worked fine without port forwarding in place though none of the internet connections I used had CGNAT with it's overly short NAT session timeouts.
 
I have a number that is porting in to A&A, so it isn't active yet. In the control centre it has options for DTMF. The number I already had for testing, which still works, and was given to me by A&A, doesn't have this option.

I asked A&A and it took a while for them to work out what I was saying. They kept saying the option will appear when the number is ported over. Eventually they realised what I was saying, which is the option is shown on the non active porting in line and NOT the one I've been using for a few weeks.

Anyway, can anyone help me with why it is there and what it does, as again A&A didn't know?

Also when I switched my login details to the number being ported, it wont authorise, even though the account appears active. Is this correct? I need my softphone app to start working the moment the number moves over, and I'm worried something isn't not right. A&A said it will authorise once the port completes, but as they haven't been great with tech support so far I'm wary.

DTMF.jpg
 
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Is the account number for your test number the same as that for your number that's porting in? If it's different then I wouldn't be surprised that the account with the porting number isn't active yet.

When I've ported numbers in the past I've had a temporary number which was removed at the point the other number ported in. It was the same account and therefore login details for the test & porting number.
 
I have a number that is porting in to A&A, so it isn't active yet. In the control centre it has options for DTMF. The number I already had for testing, which still works, and was given to me by A&A, doesn't have this option.
do you have local phones(eg dect) that are relying on receiving dialling tones for particular functionality like remotely listening to messages
(was going to ask , anyway, if A&A setup permits leaving messages via phones you have connected to grandstand, which would be important to me ... or A&A will send messages via email)
I would have thought grandstand would prefer the digital rfc2833 codes, can you experiment on test account.

SIP should be fine too. The Gradwell and Sipgate services that I've been using for 12 years or so worked fine
they are for business though ? bit expensive for domestic occasional use ?
 
they are for business though ? bit expensive for domestic occasional use ?

I can't remember how much the Gradwell account cost, it was a business number I had and the costs just hit my company credit card. Sipgate used to have a free account, they don't offer it any longer but I've had my account for about 10 years so I still have it. Monthly cost = £0, I only pay for calls. I added £20 credit about 9 years ago and there's still > £17 on there.
 
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