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Author Topic: Inbound SIP services  (Read 8125 times)

npr

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Re: Inbound SIP services
« Reply #15 on: April 08, 2016, 04:26:02 PM »

Sorry, I can't remember what cables came with it. I've got a collection of phone / modem cables so it wasn't a problem for me.

It's taken me considerably less than a year to recoup the cost of the box:

I was paying 6 /month for any time calls + around 5 / month on calls to mobiles, say 11 per month.
In 8 months that's 88

Now paying 9.50 once every 4 months which gives free land line calls for 120 days and calls to mobiles at 1p / minute which comes out of the 9.50.
http://www.freevoipdeal.com/calling_rates/
Note: these Betamax prices are subject to change -- need to keep track of which reseller is cheapest when it's time to renew.
In 8 months that's 19.   :cool:

Cost of box <40

The basic settings are obvious, but the VoIP settings can be a steep learning curve.
The two forums I've linked too are very helpful however.
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dragon2611

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Re: Inbound SIP services
« Reply #16 on: April 11, 2016, 11:06:49 AM »

Cheer.  As it happens Sipgate reversed their decision, so it's not clear what options would have been available, for example whether their numbers could have been ported out.   A&A looks good at first glance, however I note they don't support their SIP service if you use NAT.   My Gigaset phone works perfectly with SIPgate and I've tested with "discountvoip" and that seems to work as well, although they don't do inbound.   

Also looking at "localphone.com" which is another prepaid service.  I like the idea of pre-paid because that reduces toll fraud risk.

Pretty sure I've used AAISP from behind NAT before, I suspect they'll try to help you regardless but if it turns out it's not working due to your broken NAT device then they're not going to want to waste hours of time trying to fix/work around it.  (Some NAT routers really badly mangle SIP, they try to be clever by re-writing addresses but some implementations of the SIP ALG leave a lot to be desired.)
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Weaver

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Re: Inbound SIP services
« Reply #17 on: April 11, 2016, 08:02:20 PM »

If you use A&A then one way out of NAT misery is to go IPv6. You will then need the right VoIP kit, and I notice that A&A recommend Snom VoIP hardware which speaks IPv6.

I use A & A's VoIP service with a Siemens N300 VoIP box over IPv4 but with no NAT at all because I have an adequate sized block of 'real' global IPv4 addresses. That's from the good days before we started to really run out.

Another note. If you use a Firebrick router, as I do, this can handle the IPv4 NAT case for you because it acts as a back-to-back VoIP gateway.
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aesmith

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Re: Inbound SIP services
« Reply #18 on: April 12, 2016, 08:29:51 PM »

  ....  one way out of NAT misery is to go IPv6. ...

I can't help thinking that this "NAT is evil" is a bit of a hobby horse from A&A, not strictly borne out by the presumably millions of users currently accessing the Internet via some form of NAT.   

In fact the other week I was working on a SIP issue, calls to Holland or Germany the caller heard first a male voice saying the call can't be completed (sent as "proceeding" early media by the way), then a female voice saying something similar, sent as if the call was connected.   The ITSP uses the Broadsoft platform, and the Broadsoft guy was not happy that the customer's firewall was re-writing the via and connection headers, he wanted this "ALG" as he called it disabled.  However disabling that function broke everything - no early media for any calls which means not just no call proceeding messages but no ringback either.   Moving on we put the customer's gateway on a DMZ to remove all NAT from the picture, and it behaves exactly as it did via NAT and the much hated "ALG" - perfect for all calls except outgoing to Holland or Germany. 

By the way, in an IPv6 only world, is it expected that everyone will have PI address space?   What I mean is that its simple for me at home to address my stuff using the A&A provided addressing and avoid NAT, but what about an enterprise network with multiple sites and many different Internet connections from different ISPs?   (Off topic a bit, but the more I look at IPv6 the more I realise how different it is from what I'm used to) 
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Weaver

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Re: Inbound SIP services
« Reply #19 on: April 12, 2016, 08:46:03 PM »

I'm sure that a lot of organisations will want to get themselves organised with some PI space.
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ktz392837

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Re: Inbound SIP services
« Reply #20 on: April 22, 2016, 09:15:24 AM »

Quote from: npr
Now paying 9.50 once every 4 months which gives free land line calls for 120 days and calls to mobiles at 1p / minute which comes out of the 9.50.
http://www.freevoipdeal.com/calling_rates
  Could you share/export your working config for the obi110 and freevoipdeal (excluding passwords!)? Thanks

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npr

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Re: Inbound SIP services
« Reply #21 on: April 22, 2016, 11:05:33 AM »

Started by loading the UK-OBi110-Profile-21JUL15 from here:

https://www.ukvoipforums.com/viewtopic.php?f=25&t=368

IIRC all settings are default except the following.

Quote
System management > Auto provisioning : disable all

service providers > itsp profile A > general:

Digitmap: (116000|116111|116123|0[15]xxxxxxxxx?|0[27]x xxxx xxxx|0800xxx xxxx?|0808xxx xxxx|08001111|08[47]x xxx xxxx|0845464x|03xx xxx xxxx|!118x.|100|155|195|!09xx xxx xxxx|00xxx.|xx.|+xx.|(Mipd)|[^*#]@@.)

Name : freevoipdeal

url : www.freevoipdeal.com

service providers > itsp profile A > sip:

Proxyserver: sip:freevoipdeal.com

proxyserverport: 5060

proxyservertransport: UDP

registrarserver:  sip:freevoipdeal.com

registraserverport:  5060

Voice service > SP1 service:

Enable : ticked

Authusername: freevoipdeal username

Authpassword: freevoipdeal password

Physical Interfaces > phone port :


Digitmap: ([1-9]x?*(Mpli)|[1-9]S9|[1-9][0-9]S9|999|111|101|1471|17070|**0|***|#|**1(Msp1)|**2(Msp2)|**8(Mli)|**9(Mpp)|(Mpli))

Outboundcallroute: {([1-9]x?*(Mpli)):pp},{(<#:>|999|111|101|1471|17070):li},{**0:aa},{***:aa2},{(<**1:>(Msp1)):sp1},{(<**2:>(Msp2)):sp2},{(<**8:>(Mli)):li},{(<**9:>(Mpp)):pp},{(Mpli):pli}

PrimaryLine: SP1 Service

You may need to mess around with the "channelTXgain and channelRXgain" settings to get the voice sound levels right. Mine are both set to -1. Adjust these gain settings first before changing the same settings in the following LinePort section, This is because the settings here changes the volume for both voip and BT voice, the following only changes the volume for BT calls. Only make small changes or you will get distortion and/or echo.

Physical Interfaces > Line Port:

Digitmap: (999|112|101|111|116000|116111|116123|1471|17070|1571|0[1-9]xxxxxxxxxS0|0[15]xxxxxxxxx?|0[27]x xxxx xxxx|0800xxx xxxx?|0808xxx xxxx|08001111|08[47]x xxx xxxx|0845464x|03xx xxx xxxx|118xxx|100|155|195|09xx xxx xxxx|[2-8]xxxx.|9[0-8]xxx.|99[^9]xx.|00xxx.|xx.)

Again, you may need to mess around with the "channelTXgain and channelRXgain" settings here. Mine are still at the default 0 and 5.

Good luck, please let us know how you get on.




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ktz392837

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Re: Inbound SIP services
« Reply #22 on: April 22, 2016, 01:24:36 PM »

Great thanks that helps.  I am currently crafting my rules at the moment.  Just need to get in my head why there appears to be a lot of duplication of virtually the same codes.

Does the !ban option (eg your premium 09x rule) play an appropriate message? 

Do you know why 141 (withhold number) is not in any of the example rules?

How does dialing work do all calls need 0044 prefixing?  Does the VoIP provider sort out adding STD for local calls?

I want to ban international calls will a (!00) be enough?

Thanks
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npr

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Re: Inbound SIP services
« Reply #23 on: April 22, 2016, 03:51:04 PM »

No, the !ban doesn't play a message.

I have no experience of using 141.

Don't need to prefix the dial number with 0044 provided your IP address is registered to the UK.

At one time I had on a plusnet IP address register to the US, that caused problems so I moved to a static UK IP address which fixed the problem. Since moved internet provider to BT, dynamic IP, but and have no repeats of this problem.

You need to dial the STD code. Although you could create a digitmap to automatically add your local STD code to local numbers.

I assume you've seen the link I posted for screening withheld numbers.

To screen withheld and international calls you can change that digitmap to: {(?|00xx.):AA},{ph}

? matches number withheld, 00xx. matches international.

Alternatively the following will screen ? and silently block 00xx.

{(?):AA},{(00xx.):},{ph}

HH
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ktz392837

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Re: Inbound SIP services
« Reply #24 on: April 24, 2016, 09:25:22 AM »

Gradually building up my config and now trying to get AA setup.

Do you know why the digit code includes both 0 and 1?

What currency do you pay in with freevoipdeal.com when you topup? Do they add a 30% surcharge?

Thanks for help.
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npr

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Re: Inbound SIP services
« Reply #25 on: April 24, 2016, 12:28:38 PM »

For the screening config, you would expect it to need a digitmap of 1. It actually need 0 not 1, don't know why haven't bothered to find out why, just left it as 1|0 -- don't see it as a issue.

It's billed in Euros: 10 + admin charge + VAT = 12.5 Euros

Last time I paid that worked out at 9.50.

Don't forget after the 120 days of free landline calls you don't need to immediately topup, you start using any credit remaining at approx 0.01 Euro per minute.

Edit:
1 Euro corrected to 0.01Euro.  :-[
« Last Edit: April 24, 2016, 09:28:42 PM by npr »
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ktz392837

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Re: Inbound SIP services
« Reply #26 on: April 26, 2016, 10:09:54 AM »

AA is now working but trying to satisfy myself that it is secure.  After AA if you ***0 and other various numbers can get a response but nothing seems to allow any actual calls.  I have also set the goodbye message to user2 so it does not announce what device is being used just in case.

Going to get sip working next.  Do I need to punch holes in my firewall/nat?  Will the stun server sort out this without any changes to my setup?

Thanks
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npr

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Re: Inbound SIP services
« Reply #27 on: April 26, 2016, 11:50:02 AM »

I suggest you ask about security in the obi forum thread in my link, I'd be very interested to see any replies to that question.  ;)

As I see it, the AA has a digitmap of (0|1) so shouldn't allow any numbers being dialled other than a single 0 or 1, can't see anyone doing much with that. Don't forget there's only 8 seconds before it either disconnects or rings the phone.

Also assigning the Obitalk service as the primary line makes me more confidence of the security aspect.

All I did was disable the routers SIP ALG, didn't bother with the STUN server, didn't need any ports forwarding.

I have enable QoS in the router for the VoIP service but how that's done is specific to your router.


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aesmith

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Re: Inbound SIP services
« Reply #28 on: April 26, 2016, 04:01:40 PM »

Going to get sip working next.  Do I need to punch holes in my firewall/nat?  Will the stun server sort out this without any changes to my setup?

STUN is a "nice to have" as it means that the SIP messages will go out fully populated with your actual external addresses.  However most service providers who support end devices like this will be wise to NAT and will look at the address from which your device actually registered and use that in preference to what's in the headers.

Likewise typical end user devices also have a few tricks up their sleeve.  For example making sure that they send outbound RTP at any time where inbound is expected, so the inbound can make it through the firewall as if it was a reply to your outbound packets.   Note that this is not a NAT specific issue.
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ktz392837

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Re: Inbound SIP services
« Reply #29 on: May 28, 2016, 12:47:18 AM »

As I see it, the AA has a digitmap of (0|1) so shouldn't allow any numbers being dialled other than a single 0 or 1, can't see anyone doing much with that. Don't forget there's only 8 seconds before it either disconnects or rings the phone.

Also assigning the Obitalk service as the primary line makes me more confidence of the security aspect.

I have everything working and it is great apart from one thing if I call from a withheld number the aa message is played but if you do nothing after a period of time it rings the phone instead of disconnecting. Is there a setting I could of have missed?  Thanks
« Last Edit: May 28, 2016, 12:50:48 AM by ktz392837 »
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