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From: Michael K. <mic...@ip...> - 2021-09-06 22:54:11
|
Hi Group Just wondering what you would consider is the maximum number of clients for a Wireguard interface that you would feel comfortable with assuming you have enough resources to support the traffic? Im looking at connecting up to 400 remote peers. Regards Michael Knill Managing Director D: +61 2 6189 1360 P: +61 2 6140 4656 E: mic...@ip...<mailto:mic...@ip...> W: ipcsolutions.com.au<https://ipcsolutions.com.au/> [IPC Solutions] Smarter Business Communications |
|
From: Lonnie A. <li...@lo...> - 2021-09-06 00:51:05
|
Great ... be sure to test the /23 . Lonnie > On Sep 5, 2021, at 6:30 PM, Michael Knill <mic...@ip...> wrote: > > Thanks Lonnie > > No that cannot happen as the softswitch only connects to a single Astlinux peer IP address e.g. Peer 1 - 10.4.1.1/32, Peer 2 - 10.4.1.2/32 .... > All the Astlinux peers would have the same locally significant range 10.4.0.1-254. All calls to the softswitch from a remote peer are terminated by Asterisk with no direct media. > > Looks like this is what I will do then. Nice! Thanks again. > > Regards > Michael Knill > > On 6/9/21, 8:11 am, "Lonnie Abelbeck" <li...@lo...> wrote: > > That should work, be a CIDR ninja. :-) > > Though if you want your "softswitch" to route to a remote Mobile Client, /23's all around might be needed. > > Lonnie > > > >> On Sep 5, 2021, at 4:47 PM, Michael Knill <mic...@ip...> wrote: >> >> Thanks Lonnie >> >> So what I am thinking is that I will use a /23 on the remote system but continue to use /24 for my softswitch on the higher subnet. This will give a total of 250 VPN connections to the Softswitch. >> Each remote system will then have the lower subnet for local connectivity only for mobile peers and remote peers. >> >> So for your example below, the softswitch will be on 10.4.1.254/24 for instance and the remote peer will be on 10.4.1.1-250 but will be configured as a /23 so it has all 10.4.0.x for local connections. >> >> What do you think? >> >> Regards >> Michael Knill >> >> On 4/9/21, 12:35 pm, "Lonnie Abelbeck" <li...@lo...> wrote: >> >> Hi Michael, >> >> As per the docs, the range of .101 to .199 is reserved for mobile clients. >> -- >> Note -> Mobile Clients are automatically assigned a unique IP address in the range of .101 to .199 for the last octet (example here: 10.4.0.101 to 10.4.0.199). Best practice is to refrain from using IP's in this range for both this tunnel's “IPv4 Address” (above) and Remote Peer's IP address so both configuration types can coexist. Similarly for IPv6 the Mobile Client reserved range is …:0101 to …:0199. >> -- >> When a new Mobile Client is added, it will only check other mobile clients for uniqueness, not manually added remote peers. >> >> >> Alternatively, if you need more than ~150 manually added remote peers, it should be possible to use a /23 (255.255.254.0) IPv4 NetMask. >> >> Using: netcalc 10.4.0.1/23 >> -- >> HostMin : 10.4.0.1 00001010.00000100.0000000 0.00000001 >> HostMax : 10.4.1.254 00001010.00000100.0000000 1.11111110 >> -- >> Here the reserved mobile client range is still 10.4.0.101 to 10.4.0.199 >> >> You have the previous ~150 manually added remote peer range plus a ~250 10.4.1.x range. >> >> This /23 subnet should work for the WireGuard -> Tunnel Options: -> IPv4 NetMask: 255.255.254.0 >> >> but I have not tested it much. Would that work for you? >> >> Lonnie >> >> >> >>> On Sep 3, 2021, at 7:46 PM, Michael Knill <mic...@ip...> wrote: >>> >>> Hi Group >>> >>> Is there any reason that I could not use the .101 to .199 subnet addresses for Remote Peers? If I do add a mobile peer will it check Remote Peers when allocating an IP addresses or would I need to manually check there are no duplicates? >>> As I am moving to cloud hosting most of my systems now with direct mobile connectivity, I don't need to use mobile peers but I do need the address space. >>> >>> Regards >>> >>> Michael Knill >>> Managing Director >>> >>> D: +61 2 6189 1360 >>> P: +61 2 6140 4656 >>> E: mic...@ip... >>> W: ipcsolutions.com.au >>> >>> <image001.png> >>> Smarter Business Communications >>> >>> _______________________________________________ >>> Astlinux-users mailing list >>> Ast...@li... >>> https://lists.sourceforge.net/lists/listinfo/astlinux-users >>> >>> Donations to support AstLinux are graciously accepted via PayPal to pa...@kr.... >> >> >> >> _______________________________________________ >> Astlinux-users mailing list >> Ast...@li... >> https://lists.sourceforge.net/lists/listinfo/astlinux-users >> >> Donations to support AstLinux are graciously accepted via PayPal to pa...@kr.... >> >> >> _______________________________________________ >> Astlinux-users mailing list >> Ast...@li... >> https://lists.sourceforge.net/lists/listinfo/astlinux-users >> >> Donations to support AstLinux are graciously accepted via PayPal to pa...@kr.... > > > > _______________________________________________ > Astlinux-users mailing list > Ast...@li... > https://lists.sourceforge.net/lists/listinfo/astlinux-users > > Donations to support AstLinux are graciously accepted via PayPal to pa...@kr.... > > > _______________________________________________ > Astlinux-users mailing list > Ast...@li... > https://lists.sourceforge.net/lists/listinfo/astlinux-users > > Donations to support AstLinux are graciously accepted via PayPal to pa...@kr.... |
|
From: Michael K. <mic...@ip...> - 2021-09-05 23:30:20
|
Thanks Lonnie
No that cannot happen as the softswitch only connects to a single Astlinux peer IP address e.g. Peer 1 - 10.4.1.1/32, Peer 2 - 10.4.1.2/32 ....
All the Astlinux peers would have the same locally significant range 10.4.0.1-254. All calls to the softswitch from a remote peer are terminated by Asterisk with no direct media.
Looks like this is what I will do then. Nice! Thanks again.
Regards
Michael Knill
On 6/9/21, 8:11 am, "Lonnie Abelbeck" <li...@lo...> wrote:
That should work, be a CIDR ninja. :-)
Though if you want your "softswitch" to route to a remote Mobile Client, /23's all around might be needed.
Lonnie
> On Sep 5, 2021, at 4:47 PM, Michael Knill <mic...@ip...> wrote:
>
> Thanks Lonnie
>
> So what I am thinking is that I will use a /23 on the remote system but continue to use /24 for my softswitch on the higher subnet. This will give a total of 250 VPN connections to the Softswitch.
> Each remote system will then have the lower subnet for local connectivity only for mobile peers and remote peers.
>
> So for your example below, the softswitch will be on 10.4.1.254/24 for instance and the remote peer will be on 10.4.1.1-250 but will be configured as a /23 so it has all 10.4.0.x for local connections.
>
> What do you think?
>
> Regards
> Michael Knill
>
> On 4/9/21, 12:35 pm, "Lonnie Abelbeck" <li...@lo...> wrote:
>
> Hi Michael,
>
> As per the docs, the range of .101 to .199 is reserved for mobile clients.
> --
> Note -> Mobile Clients are automatically assigned a unique IP address in the range of .101 to .199 for the last octet (example here: 10.4.0.101 to 10.4.0.199). Best practice is to refrain from using IP's in this range for both this tunnel's “IPv4 Address” (above) and Remote Peer's IP address so both configuration types can coexist. Similarly for IPv6 the Mobile Client reserved range is …:0101 to …:0199.
> --
> When a new Mobile Client is added, it will only check other mobile clients for uniqueness, not manually added remote peers.
>
>
> Alternatively, if you need more than ~150 manually added remote peers, it should be possible to use a /23 (255.255.254.0) IPv4 NetMask.
>
> Using: netcalc 10.4.0.1/23
> --
> HostMin : 10.4.0.1 00001010.00000100.0000000 0.00000001
> HostMax : 10.4.1.254 00001010.00000100.0000000 1.11111110
> --
> Here the reserved mobile client range is still 10.4.0.101 to 10.4.0.199
>
> You have the previous ~150 manually added remote peer range plus a ~250 10.4.1.x range.
>
> This /23 subnet should work for the WireGuard -> Tunnel Options: -> IPv4 NetMask: 255.255.254.0
>
> but I have not tested it much. Would that work for you?
>
> Lonnie
>
>
>
>> On Sep 3, 2021, at 7:46 PM, Michael Knill <mic...@ip...> wrote:
>>
>> Hi Group
>>
>> Is there any reason that I could not use the .101 to .199 subnet addresses for Remote Peers? If I do add a mobile peer will it check Remote Peers when allocating an IP addresses or would I need to manually check there are no duplicates?
>> As I am moving to cloud hosting most of my systems now with direct mobile connectivity, I don't need to use mobile peers but I do need the address space.
>>
>> Regards
>>
>> Michael Knill
>> Managing Director
>>
>> D: +61 2 6189 1360
>> P: +61 2 6140 4656
>> E: mic...@ip...
>> W: ipcsolutions.com.au
>>
>> <image001.png>
>> Smarter Business Communications
>>
>> _______________________________________________
>> Astlinux-users mailing list
>> Ast...@li...
>> https://lists.sourceforge.net/lists/listinfo/astlinux-users
>>
>> Donations to support AstLinux are graciously accepted via PayPal to pa...@kr....
>
>
>
> _______________________________________________
> Astlinux-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/astlinux-users
>
> Donations to support AstLinux are graciously accepted via PayPal to pa...@kr....
>
>
> _______________________________________________
> Astlinux-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/astlinux-users
>
> Donations to support AstLinux are graciously accepted via PayPal to pa...@kr....
_______________________________________________
Astlinux-users mailing list
Ast...@li...
https://lists.sourceforge.net/lists/listinfo/astlinux-users
Donations to support AstLinux are graciously accepted via PayPal to pa...@kr....
|
|
From: Lonnie A. <li...@lo...> - 2021-09-05 22:10:55
|
That should work, be a CIDR ninja. :-) Though if you want your "softswitch" to route to a remote Mobile Client, /23's all around might be needed. Lonnie > On Sep 5, 2021, at 4:47 PM, Michael Knill <mic...@ip...> wrote: > > Thanks Lonnie > > So what I am thinking is that I will use a /23 on the remote system but continue to use /24 for my softswitch on the higher subnet. This will give a total of 250 VPN connections to the Softswitch. > Each remote system will then have the lower subnet for local connectivity only for mobile peers and remote peers. > > So for your example below, the softswitch will be on 10.4.1.254/24 for instance and the remote peer will be on 10.4.1.1-250 but will be configured as a /23 so it has all 10.4.0.x for local connections. > > What do you think? > > Regards > Michael Knill > > On 4/9/21, 12:35 pm, "Lonnie Abelbeck" <li...@lo...> wrote: > > Hi Michael, > > As per the docs, the range of .101 to .199 is reserved for mobile clients. > -- > Note -> Mobile Clients are automatically assigned a unique IP address in the range of .101 to .199 for the last octet (example here: 10.4.0.101 to 10.4.0.199). Best practice is to refrain from using IP's in this range for both this tunnel's “IPv4 Address” (above) and Remote Peer's IP address so both configuration types can coexist. Similarly for IPv6 the Mobile Client reserved range is …:0101 to …:0199. > -- > When a new Mobile Client is added, it will only check other mobile clients for uniqueness, not manually added remote peers. > > > Alternatively, if you need more than ~150 manually added remote peers, it should be possible to use a /23 (255.255.254.0) IPv4 NetMask. > > Using: netcalc 10.4.0.1/23 > -- > HostMin : 10.4.0.1 00001010.00000100.0000000 0.00000001 > HostMax : 10.4.1.254 00001010.00000100.0000000 1.11111110 > -- > Here the reserved mobile client range is still 10.4.0.101 to 10.4.0.199 > > You have the previous ~150 manually added remote peer range plus a ~250 10.4.1.x range. > > This /23 subnet should work for the WireGuard -> Tunnel Options: -> IPv4 NetMask: 255.255.254.0 > > but I have not tested it much. Would that work for you? > > Lonnie > > > >> On Sep 3, 2021, at 7:46 PM, Michael Knill <mic...@ip...> wrote: >> >> Hi Group >> >> Is there any reason that I could not use the .101 to .199 subnet addresses for Remote Peers? If I do add a mobile peer will it check Remote Peers when allocating an IP addresses or would I need to manually check there are no duplicates? >> As I am moving to cloud hosting most of my systems now with direct mobile connectivity, I don't need to use mobile peers but I do need the address space. >> >> Regards >> >> Michael Knill >> Managing Director >> >> D: +61 2 6189 1360 >> P: +61 2 6140 4656 >> E: mic...@ip... >> W: ipcsolutions.com.au >> >> <image001.png> >> Smarter Business Communications >> >> _______________________________________________ >> Astlinux-users mailing list >> Ast...@li... >> https://lists.sourceforge.net/lists/listinfo/astlinux-users >> >> Donations to support AstLinux are graciously accepted via PayPal to pa...@kr.... > > > > _______________________________________________ > Astlinux-users mailing list > Ast...@li... > https://lists.sourceforge.net/lists/listinfo/astlinux-users > > Donations to support AstLinux are graciously accepted via PayPal to pa...@kr.... > > > _______________________________________________ > Astlinux-users mailing list > Ast...@li... > https://lists.sourceforge.net/lists/listinfo/astlinux-users > > Donations to support AstLinux are graciously accepted via PayPal to pa...@kr.... |
|
From: Michael K. <mic...@ip...> - 2021-09-05 21:47:25
|
Thanks Lonnie
So what I am thinking is that I will use a /23 on the remote system but continue to use /24 for my softswitch on the higher subnet. This will give a total of 250 VPN connections to the Softswitch.
Each remote system will then have the lower subnet for local connectivity only for mobile peers and remote peers.
So for your example below, the softswitch will be on 10.4.1.254/24 for instance and the remote peer will be on 10.4.1.1-250 but will be configured as a /23 so it has all 10.4.0.x for local connections.
What do you think?
Regards
Michael Knill
On 4/9/21, 12:35 pm, "Lonnie Abelbeck" <li...@lo...> wrote:
Hi Michael,
As per the docs, the range of .101 to .199 is reserved for mobile clients.
--
Note -> Mobile Clients are automatically assigned a unique IP address in the range of .101 to .199 for the last octet (example here: 10.4.0.101 to 10.4.0.199). Best practice is to refrain from using IP's in this range for both this tunnel's “IPv4 Address” (above) and Remote Peer's IP address so both configuration types can coexist. Similarly for IPv6 the Mobile Client reserved range is …:0101 to …:0199.
--
When a new Mobile Client is added, it will only check other mobile clients for uniqueness, not manually added remote peers.
Alternatively, if you need more than ~150 manually added remote peers, it should be possible to use a /23 (255.255.254.0) IPv4 NetMask.
Using: netcalc 10.4.0.1/23
--
HostMin : 10.4.0.1 00001010.00000100.0000000 0.00000001
HostMax : 10.4.1.254 00001010.00000100.0000000 1.11111110
--
Here the reserved mobile client range is still 10.4.0.101 to 10.4.0.199
You have the previous ~150 manually added remote peer range plus a ~250 10.4.1.x range.
This /23 subnet should work for the WireGuard -> Tunnel Options: -> IPv4 NetMask: 255.255.254.0
but I have not tested it much. Would that work for you?
Lonnie
> On Sep 3, 2021, at 7:46 PM, Michael Knill <mic...@ip...> wrote:
>
> Hi Group
>
> Is there any reason that I could not use the .101 to .199 subnet addresses for Remote Peers? If I do add a mobile peer will it check Remote Peers when allocating an IP addresses or would I need to manually check there are no duplicates?
> As I am moving to cloud hosting most of my systems now with direct mobile connectivity, I don't need to use mobile peers but I do need the address space.
>
> Regards
>
> Michael Knill
> Managing Director
>
> D: +61 2 6189 1360
> P: +61 2 6140 4656
> E: mic...@ip...
> W: ipcsolutions.com.au
>
> <image001.png>
> Smarter Business Communications
>
> _______________________________________________
> Astlinux-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/astlinux-users
>
> Donations to support AstLinux are graciously accepted via PayPal to pa...@kr....
_______________________________________________
Astlinux-users mailing list
Ast...@li...
https://lists.sourceforge.net/lists/listinfo/astlinux-users
Donations to support AstLinux are graciously accepted via PayPal to pa...@kr....
|
|
From: Lonnie A. <li...@lo...> - 2021-09-04 02:34:53
|
Hi Michael, As per the docs, the range of .101 to .199 is reserved for mobile clients. -- Note -> Mobile Clients are automatically assigned a unique IP address in the range of .101 to .199 for the last octet (example here: 10.4.0.101 to 10.4.0.199). Best practice is to refrain from using IP's in this range for both this tunnel's “IPv4 Address” (above) and Remote Peer's IP address so both configuration types can coexist. Similarly for IPv6 the Mobile Client reserved range is …:0101 to …:0199. -- When a new Mobile Client is added, it will only check other mobile clients for uniqueness, not manually added remote peers. Alternatively, if you need more than ~150 manually added remote peers, it should be possible to use a /23 (255.255.254.0) IPv4 NetMask. Using: netcalc 10.4.0.1/23 -- HostMin : 10.4.0.1 00001010.00000100.0000000 0.00000001 HostMax : 10.4.1.254 00001010.00000100.0000000 1.11111110 -- Here the reserved mobile client range is still 10.4.0.101 to 10.4.0.199 You have the previous ~150 manually added remote peer range plus a ~250 10.4.1.x range. This /23 subnet should work for the WireGuard -> Tunnel Options: -> IPv4 NetMask: 255.255.254.0 but I have not tested it much. Would that work for you? Lonnie > On Sep 3, 2021, at 7:46 PM, Michael Knill <mic...@ip...> wrote: > > Hi Group > > Is there any reason that I could not use the .101 to .199 subnet addresses for Remote Peers? If I do add a mobile peer will it check Remote Peers when allocating an IP addresses or would I need to manually check there are no duplicates? > As I am moving to cloud hosting most of my systems now with direct mobile connectivity, I don't need to use mobile peers but I do need the address space. > > Regards > > Michael Knill > Managing Director > > D: +61 2 6189 1360 > P: +61 2 6140 4656 > E: mic...@ip... > W: ipcsolutions.com.au > > <image001.png> > Smarter Business Communications > > _______________________________________________ > Astlinux-users mailing list > Ast...@li... > https://lists.sourceforge.net/lists/listinfo/astlinux-users > > Donations to support AstLinux are graciously accepted via PayPal to pa...@kr.... |
|
From: Michael K. <mic...@ip...> - 2021-09-04 00:46:19
|
Hi Group Is there any reason that I could not use the .101 to .199 subnet addresses for Remote Peers? If I do add a mobile peer will it check Remote Peers when allocating an IP addresses or would I need to manually check there are no duplicates? As I am moving to cloud hosting most of my systems now with direct mobile connectivity, I don't need to use mobile peers but I do need the address space. Regards Michael Knill Managing Director D: +61 2 6189 1360 P: +61 2 6140 4656 E: mic...@ip...<mailto:mic...@ip...> W: ipcsolutions.com.au<https://ipcsolutions.com.au/> [IPC Solutions] Smarter Business Communications |
|
From: David K. <da...@ke...> - 2021-08-29 21:10:41
|
Thanks to everyone for the suggestions. I have signed up for voip.ms and am now routing outbound calls through them. I'll decide what to do about inbound DID once I burn down my pre-pay at Vitelity, that will take a few months as it just did an auto top-up. So far I have been impressed with voip.ms, the call quality has been excellent. Interestingly they support (beta) G.722 and so I turned that on and it is using it... which means no transcoding on my Astlinux box from my Yealink phones through to voip.ms. I have no idea if their connection to e.g. US mobile carriers is also HD voice or not, but call quality to my cell phone is very good (I am T-Mobile). To unlock all international destinations you have to provide identity info. I have not done that yet as most everywhere I am likely to call is available without that. But oddly, calls to Australia are blocked by default and while rare for me to need to call there, I feel I need to unlock it "just in case". The only downside so far is that some of their rates to international mobiles and "toll free" can be very high, and highly variable as their rates are very granular, for example first four digits of a UK mobile phone number will determine the rate and can vary from 2 cents (my mothers mobile phone) to 58 cents (my sister's mobile phone). That is using their "premium" routes, but even the value route is still 18 cents. There is no logic to it, it's not like one mobile carrier is in one price bracket and another in a different one, it's whatever your first four digits are (number portability means that there is no longer any correlation between carriers and number prefixes). So care is required if calling a non-land line international (which for many countries has always been the case). And don't even think about calling a UK toll free ($1.50 a minute) or 900 number ($13.40 a minute). AnveoDirect would solve the rate problem for UK mobiles/toll free. Their prices are under 2 cents for all UK mobile, which is absolutely amazing (that is their prime rate). Their consumer retail rates for UK mobile however are worse than voip.ms which is really odd given how low their bulk/direct rates are. But I would have to deposit $35 with them to get started and I'm not sure that I make enough calls to UK mobiles to make it worthwhile. David On Mon, Aug 23, 2021 at 11:36 AM Lonnie Abelbeck <li...@lo...> wrote: > Thanks Dan for your SIP provider experiences. Good info. > > I would like to see some SIP provider partner with Vultr or Linode to > offer a direct connection to their SIP/RTP endpoints, then AstLinux could > be spun-up offering WireGuard access to the SIP/RTP. Exposing SIP/RTP to > the public would only be an option and could be firewalled by AstLinux. > > But as usual, the devil is in the detail. > > Lonnie > > > > On Aug 23, 2021, at 7:31 AM, Dan Ryson <da...@ry...> wrote: > > > > I'm sure you've looked into this but when I hear of problems precisely > at 15 minutes, failed SIP re-invites come to mind. If so, it might be > worth exploring whether a SIP re-invite was sent but not acknowledged for > some reason. > > > > I was glad to hear this question regarding providers. Since it has > never come up, I presumed it was a taboo topic. As a small-time hobbyist > user, my experience is limited in scope but many years have been spent on a > quest for an ideal AstLinux fit. I've had varying degrees of success and > many disappointments. > > > > As AstLinux is so feature-rich, it can largely stand on its own with > minimal help from providers. As a result, I've gravitated to wholesale > providers that offer direct RTP from carriers, which typically reduces > latency, improves voice quality, and lowers cost. Although reduced cost > wasn't my primary motivator, it does pay for a $5/month Linode AstLinux > droplet that's used as a static-IP, SIP-only, proxy server and failover > backup. [Note: With many wholesale providers, you're on your own to handle > calls during a PBX outage.] > > > > Here are a few candidates that you may wish to explore and test. Use at > your own risk! > > • Voxbeam: Used seven years for US domestic termination. I have > limited experience with their DIDs. International termination available. > UK owned and operated so may be particularly useful for calls to western > Europe. E911 is provided only by their retail product "Localphone," which > has a bit more polish and may be more akin to Vitelity. Offers both > "direct" RTP and via points of presence in US and Amsterdam. Listed first > because of your specific requirements regarding UK calls. > > • AnveoDirect: Used nine years for US domestic DIDs but only > recently for US domestic termination. Some international termination > available. It is possible to select among multiple termination carriers or > use their unique least-cost-routing. Does not support registration for > DIDs but I use a static IP. Registration and E911 is provided only by > their retail product "Anveo," which I don't know much about. > > • BulkVS: Six months experience but my early experience with DIDs > and US domestic termination has been good. At the very least, it's an > intriguing product worthy of review for US domestic-only use. Offers E911, > US domestic termination only. Direct RTP only. Supports registration as > well as IP authentication. Listed last only because it doesn't provide UK > support. > > One factor to consider when selecting termination carriers within the US > is the FCC's recent push for STIR/SHAKEN compliance (more) that adds SIP > header signatures to facilitate CallerID authentication. From a practical > standpoint in the short term, STIR/SHAKEN compliance for "little guys" like > me requires use of the same termination provider as the DID provider. > > > > Dan > > > > On Sun, Aug 22, 2021 at 05:52 PM, David Kerr <da...@ke...> wrote: > > Can anyone recommend a good SIP trunk/pstn provider? I've been using > Vitelity for years and had been happy with them, but recently long distance > calls (well specifically calls from US to UK) have lost audio about 10-15 > minutes into the call. Immediately calling back sometimes works, or > sometimes it requires the other person to call me. > > > > I'm wondering if I should try a different provider. Any suggestions? > > > > Thanks > > David > > _______________________________________________ > > Astlinux-users mailing list > > Ast...@li... > > https://lists.sourceforge.net/lists/listinfo/astlinux-users > > > > Donations to support AstLinux are graciously accepted via PayPal to > pa...@kr.... > > > > _______________________________________________ > Astlinux-users mailing list > Ast...@li... > https://lists.sourceforge.net/lists/listinfo/astlinux-users > > Donations to support AstLinux are graciously accepted via PayPal to > pa...@kr.... |
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From: Michael K. <li...@mk...> - 2021-08-23 15:45:32
|
BTW: I also had the "15 minutes" issue with "Deutsche Telekom" SIP-trunks here in Germany, and I could solve this by setting "session-timers=refuse" in "sip.conf" in those cases (default is "accept"). > Am 23.08.2021 um 14:31 schrieb Dan Ryson <da...@ry...>: > > I'm sure you've looked into this but when I hear of problems precisely at 15 minutes, failed SIP re-invites come to mind. If so, it might be worth exploring whether a SIP re-invite was sent but not acknowledged for some reason. > > I was glad to hear this question regarding providers. Since it has never come up, I presumed it was a taboo topic. As a small-time hobbyist user, my experience is limited in scope but many years have been spent on a quest for an ideal AstLinux fit. I've had varying degrees of success and many disappointments. > > As AstLinux is so feature-rich, it can largely stand on its own with minimal help from providers. As a result, I've gravitated to wholesale providers that offer direct RTP from carriers, which typically reduces latency, improves voice quality, and lowers cost. Although reduced cost wasn't my primary motivator, it does pay for a $5/month Linode AstLinux droplet that's used as a static-IP, SIP-only, proxy server and failover backup. [Note: With many wholesale providers, you're on your own to handle calls during a PBX outage.] > > Here are a few candidates that you may wish to explore and test. Use at your own risk! > • Voxbeam: Used seven years for US domestic termination. I have limited experience with their DIDs. International termination available. UK owned and operated so may be particularly useful for calls to western Europe. E911 is provided only by their retail product "Localphone," which has a bit more polish and may be more akin to Vitelity. Offers both "direct" RTP and via points of presence in US and Amsterdam. Listed first because of your specific requirements regarding UK calls. > • AnveoDirect: Used nine years for US domestic DIDs but only recently for US domestic termination. Some international termination available. It is possible to select among multiple termination carriers or use their unique least-cost-routing. Does not support registration for DIDs but I use a static IP. Registration and E911 is provided only by their retail product "Anveo," which I don't know much about. > • BulkVS: Six months experience but my early experience with DIDs and US domestic termination has been good. At the very least, it's an intriguing product worthy of review for US domestic-only use. Offers E911, US domestic termination only. Direct RTP only. Supports registration as well as IP authentication. Listed last only because it doesn't provide UK support. > One factor to consider when selecting termination carriers within the US is the FCC's recent push for STIR/SHAKEN compliance (more) that adds SIP header signatures to facilitate CallerID authentication. From a practical standpoint in the short term, STIR/SHAKEN compliance for "little guys" like me requires use of the same termination provider as the DID provider. > > Dan > > On Sun, Aug 22, 2021 at 05:52 PM, David Kerr <da...@ke...> wrote: > Can anyone recommend a good SIP trunk/pstn provider? I've been using Vitelity for years and had been happy with them, but recently long distance calls (well specifically calls from US to UK) have lost audio about 10-15 minutes into the call. Immediately calling back sometimes works, or sometimes it requires the other person to call me. > > I'm wondering if I should try a different provider. Any suggestions? > > Thanks > David Michael http://www.mksolutions.info |
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From: Lonnie A. <li...@lo...> - 2021-08-23 15:36:12
|
Thanks Dan for your SIP provider experiences. Good info. I would like to see some SIP provider partner with Vultr or Linode to offer a direct connection to their SIP/RTP endpoints, then AstLinux could be spun-up offering WireGuard access to the SIP/RTP. Exposing SIP/RTP to the public would only be an option and could be firewalled by AstLinux. But as usual, the devil is in the detail. Lonnie > On Aug 23, 2021, at 7:31 AM, Dan Ryson <da...@ry...> wrote: > > I'm sure you've looked into this but when I hear of problems precisely at 15 minutes, failed SIP re-invites come to mind. If so, it might be worth exploring whether a SIP re-invite was sent but not acknowledged for some reason. > > I was glad to hear this question regarding providers. Since it has never come up, I presumed it was a taboo topic. As a small-time hobbyist user, my experience is limited in scope but many years have been spent on a quest for an ideal AstLinux fit. I've had varying degrees of success and many disappointments. > > As AstLinux is so feature-rich, it can largely stand on its own with minimal help from providers. As a result, I've gravitated to wholesale providers that offer direct RTP from carriers, which typically reduces latency, improves voice quality, and lowers cost. Although reduced cost wasn't my primary motivator, it does pay for a $5/month Linode AstLinux droplet that's used as a static-IP, SIP-only, proxy server and failover backup. [Note: With many wholesale providers, you're on your own to handle calls during a PBX outage.] > > Here are a few candidates that you may wish to explore and test. Use at your own risk! > • Voxbeam: Used seven years for US domestic termination. I have limited experience with their DIDs. International termination available. UK owned and operated so may be particularly useful for calls to western Europe. E911 is provided only by their retail product "Localphone," which has a bit more polish and may be more akin to Vitelity. Offers both "direct" RTP and via points of presence in US and Amsterdam. Listed first because of your specific requirements regarding UK calls. > • AnveoDirect: Used nine years for US domestic DIDs but only recently for US domestic termination. Some international termination available. It is possible to select among multiple termination carriers or use their unique least-cost-routing. Does not support registration for DIDs but I use a static IP. Registration and E911 is provided only by their retail product "Anveo," which I don't know much about. > • BulkVS: Six months experience but my early experience with DIDs and US domestic termination has been good. At the very least, it's an intriguing product worthy of review for US domestic-only use. Offers E911, US domestic termination only. Direct RTP only. Supports registration as well as IP authentication. Listed last only because it doesn't provide UK support. > One factor to consider when selecting termination carriers within the US is the FCC's recent push for STIR/SHAKEN compliance (more) that adds SIP header signatures to facilitate CallerID authentication. From a practical standpoint in the short term, STIR/SHAKEN compliance for "little guys" like me requires use of the same termination provider as the DID provider. > > Dan > > On Sun, Aug 22, 2021 at 05:52 PM, David Kerr <da...@ke...> wrote: > Can anyone recommend a good SIP trunk/pstn provider? I've been using Vitelity for years and had been happy with them, but recently long distance calls (well specifically calls from US to UK) have lost audio about 10-15 minutes into the call. Immediately calling back sometimes works, or sometimes it requires the other person to call me. > > I'm wondering if I should try a different provider. Any suggestions? > > Thanks > David > _______________________________________________ > Astlinux-users mailing list > Ast...@li... > https://lists.sourceforge.net/lists/listinfo/astlinux-users > > Donations to support AstLinux are graciously accepted via PayPal to pa...@kr.... |
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From: Dan R. <da...@ry...> - 2021-08-23 12:58:32
|
I'm sure you've looked into this but when I hear of problems precisely at 15 minutes, failed SIP re-invites come to mind. If so, it might be worth exploring whether a SIP re-invite was sent but not acknowledged for some reason. I was glad to hear this question regarding providers. Since it has never come up, I presumed it was a taboo topic. As a small-time hobbyist user, my experience is limited in scope but many years have been spent on a quest for an ideal AstLinux fit. I've had varying degrees of success and many disappointments. As AstLinux is so feature-rich, it can largely stand on its own with minimal help from providers. As a result, I've gravitated to wholesale providers that offer direct RTP from carriers, which typically reduces latency, improves voice quality, and lowers cost. Although reduced cost wasn't my primary motivator, it does pay for a $5/month Linode AstLinux droplet that's used as a static-IP, SIP-only, proxy server and failover backup. [Note: With many wholesale providers, you're on your own to handle calls during a PBX outage.] Here are a few candidates that you may wish to explore and test. Use at your own risk! * Voxbeam (https://www.voxbeam.com/): Used seven years for US domestic termination. I have limited experience with their DIDs. International termination available. UK owned and operated so may be particularly useful for calls to western Europe. E911 is provided only by their retail product "Localphone," which has a bit more polish and may be more akin to Vitelity. Offers both "direct" RTP and via points of presence in US and Amsterdam. Listed first because of your specific requirements regarding UK calls. * AnveoDirect (http://anveodirect.com/): Used nine years for US domestic DIDs but only recently for US domestic termination. Some international termination available. It is possible to select among multiple termination carriers or use their unique least-cost-routing. Does not support registration for DIDs but I use a static IP. Registration and E911 is provided only by their retail product "Anveo," which I don't know much about. * BulkVS (https://www.bulkvs.com/): Six months experience but my early experience with DIDs and US domestic termination has been good. At the very least, it's an intriguing product worthy of review for US domestic-only use. Offers E911, US domestic termination only. Direct RTP only. Supports registration as well as IP authentication. Listed last only because it doesn't provide UK support. One factor to consider when selecting termination carriers within the US is the FCC's recent push for STIR/SHAKEN compliance (more (https://en.wikipedia.org/wiki/STIR/SHAKEN)) that adds SIP header signatures to facilitate CallerID authentication. From a practical standpoint in the short term, STIR/SHAKEN compliance for "little guys" like me requires use of the same termination provider as the DID provider. Dan On Sun, Aug 22, 2021 at 05:52 PM, David Kerr wrote: Can anyone recommend a good SIP trunk/pstn provider? I've been using Vitelity for years and had been happy with them, but recently long distance calls (well specifically calls from US to UK) have lost audio about 10-15 minutes into the call. Immediately calling back sometimes works, or sometimes it requires the other person to call me. I'm wondering if I should try a different provider. Any suggestions? ThanksDavid |
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From: Michael K. <li...@mk...> - 2021-08-23 10:36:51
|
Hi Michael, it might depend also on how many SIP users per instance are connected. For 50 or less 1 GB should be fine. For bigger systems I would keep an eye on the free RAM. > Am 23.08.2021 um 00:22 schrieb Michael Knill <mic...@ip...>: > > Thanks Lonnie. 1G it will be > > Regards > Michael Knill > > On 23/8/21, 8:11 am, "Lonnie Abelbeck" <li...@lo...> wrote: > > Hi Michael, > > Without FOP and no LXC containers, 1.0G RAM should be safe and not a worry. You could go lower, but you would have to monitor things more closely. > > Lonnie > >> On Aug 22, 2021, at 3:45 PM, Michael Knill <mic...@ip...> wrote: >> >> Hi Group >> >> I'm using VMware vCloud with one of my providers and have set up a Virtual Data Centre. I'm looking to set up a few Astlinux systems in this environment. >> Although you can overcommit on CPU, you cannot on RAM and as this is fairly expensive, I'm wanting to go as low as I am comfortable on each Astlinux system. >> >> Just wondering what the maximum RAM usage you should ever see on an Astlinux system assuming no FOP is running? I have currently made it 1.5G but I think I can go lower than this. >> >> Regards >> >> Michael Knill >> Managing Director >> >> D: +61 2 6189 1360 >> P: +61 2 6140 4656 >> E: mic...@ip... >> W: ipcsolutions.com.au >> >> <image001.png> >> Smarter Business Communications Michael http://www.mksolutions.info |
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From: John N. <jn...@co...> - 2021-08-23 02:20:39
|
I have used voip.ms for more than 10 years, and have had little to no issues with them I can't comment on their SIP performance, I use the IAX protocol into my node with no issues, they support both. They are constantly improving, adding new servers around the U.S and adding features from time to time. Over the past 10 years, their costs have been reduced as well. Though I seldom needed it, I have found their support excellent and accurate. They also have a very nice spam call blocking facility included Not sure if it is current, but from time to time they have free number porting. I only call within the U.S. and even have outbound international blocked. John Novack David Kerr wrote: > Thanks Lonnie, I'll give voip.ms <http://voip.ms> a try for outbound. If they work well I may move inbound over to them as well. > > Thanks > David > > On Sun, Aug 22, 2021 at 6:22 PM Lonnie Abelbeck <li...@lo... <mailto:li...@lo...>> wrote: > > Hi David, > > I also have used Vitelity for years, I have not noticed any issues but only call within the US. > > I have not personally used voip.ms <http://voip.ms>, but I hear good things about it https://www.voip.ms/ > > https://lawrencesystems.com/review-using-voip-ms-for-sip-cloud-and-pbx-phone-services/ > > Lonnie > > > > > On Aug 22, 2021, at 3:49 PM, David Kerr <da...@ke... <mailto:da...@ke...>> wrote: > > > > Can anyone recommend a good SIP trunk/pstn provider? I've been using Vitelity for years and had been happy with them, but recently long distance calls (well specifically calls from US to UK) have lost audio about 10-15 minutes into the call. Immediately calling back sometimes works, or sometimes it requires the other person to call me. > > > > I'm wondering if I should try a different provider. Any suggestions? > > > > Thanks > > David > > _______________________________________________ > > Astlinux-users mailing list > > Ast...@li... <mailto:Ast...@li...> > > https://lists.sourceforge.net/lists/listinfo/astlinux-users > > > > Donations to support AstLinux are graciously accepted via PayPal to pa...@kr... <mailto:pa...@kr...>. > > > > _______________________________________________ > Astlinux-users mailing list > Ast...@li... <mailto:Ast...@li...> > https://lists.sourceforge.net/lists/listinfo/astlinux-users > > Donations to support AstLinux are graciously accepted via PayPal to pa...@kr... <mailto:pa...@kr...>. > > > > _______________________________________________ > Astlinux-users mailing list > Ast...@li... > https://lists.sourceforge.net/lists/listinfo/astlinux-users > > Donations to support AstLinux are graciously accepted via PayPal to pa...@kr.... -- Dog is my Co-Pilot |
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From: David K. <da...@ke...> - 2021-08-23 02:01:57
|
Thanks Lonnie, I'll give voip.ms a try for outbound. If they work well I may move inbound over to them as well. Thanks David On Sun, Aug 22, 2021 at 6:22 PM Lonnie Abelbeck <li...@lo...> wrote: > Hi David, > > I also have used Vitelity for years, I have not noticed any issues but > only call within the US. > > I have not personally used voip.ms, but I hear good things about it > https://www.voip.ms/ > > > https://lawrencesystems.com/review-using-voip-ms-for-sip-cloud-and-pbx-phone-services/ > > Lonnie > > > > > On Aug 22, 2021, at 3:49 PM, David Kerr <da...@ke...> wrote: > > > > Can anyone recommend a good SIP trunk/pstn provider? I've been using > Vitelity for years and had been happy with them, but recently long distance > calls (well specifically calls from US to UK) have lost audio about 10-15 > minutes into the call. Immediately calling back sometimes works, or > sometimes it requires the other person to call me. > > > > I'm wondering if I should try a different provider. Any suggestions? > > > > Thanks > > David > > _______________________________________________ > > Astlinux-users mailing list > > Ast...@li... > > https://lists.sourceforge.net/lists/listinfo/astlinux-users > > > > Donations to support AstLinux are graciously accepted via PayPal to > pa...@kr.... > > > > _______________________________________________ > Astlinux-users mailing list > Ast...@li... > https://lists.sourceforge.net/lists/listinfo/astlinux-users > > Donations to support AstLinux are graciously accepted via PayPal to > pa...@kr.... > |
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From: Michael K. <mic...@ip...> - 2021-08-22 22:38:05
|
Thanks Lonnie. 1G it will be
Regards
Michael Knill
On 23/8/21, 8:11 am, "Lonnie Abelbeck" <li...@lo...> wrote:
Hi Michael,
Without FOP and no LXC containers, 1.0G RAM should be safe and not a worry. You could go lower, but you would have to monitor things more closely.
Lonnie
> On Aug 22, 2021, at 3:45 PM, Michael Knill <mic...@ip...> wrote:
>
> Hi Group
>
> I'm using VMware vCloud with one of my providers and have set up a Virtual Data Centre. I'm looking to set up a few Astlinux systems in this environment.
> Although you can overcommit on CPU, you cannot on RAM and as this is fairly expensive, I'm wanting to go as low as I am comfortable on each Astlinux system.
>
> Just wondering what the maximum RAM usage you should ever see on an Astlinux system assuming no FOP is running? I have currently made it 1.5G but I think I can go lower than this.
>
> Regards
>
> Michael Knill
> Managing Director
>
> D: +61 2 6189 1360
> P: +61 2 6140 4656
> E: mic...@ip...
> W: ipcsolutions.com.au
>
> <image001.png>
> Smarter Business Communications
>
> _______________________________________________
> Astlinux-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/astlinux-users
>
> Donations to support AstLinux are graciously accepted via PayPal to pa...@kr....
_______________________________________________
Astlinux-users mailing list
Ast...@li...
https://lists.sourceforge.net/lists/listinfo/astlinux-users
Donations to support AstLinux are graciously accepted via PayPal to pa...@kr....
|
|
From: Lonnie A. <li...@lo...> - 2021-08-22 22:22:04
|
Hi David, I also have used Vitelity for years, I have not noticed any issues but only call within the US. I have not personally used voip.ms, but I hear good things about it https://www.voip.ms/ https://lawrencesystems.com/review-using-voip-ms-for-sip-cloud-and-pbx-phone-services/ Lonnie > On Aug 22, 2021, at 3:49 PM, David Kerr <da...@ke...> wrote: > > Can anyone recommend a good SIP trunk/pstn provider? I've been using Vitelity for years and had been happy with them, but recently long distance calls (well specifically calls from US to UK) have lost audio about 10-15 minutes into the call. Immediately calling back sometimes works, or sometimes it requires the other person to call me. > > I'm wondering if I should try a different provider. Any suggestions? > > Thanks > David > _______________________________________________ > Astlinux-users mailing list > Ast...@li... > https://lists.sourceforge.net/lists/listinfo/astlinux-users > > Donations to support AstLinux are graciously accepted via PayPal to pa...@kr.... |
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From: Lonnie A. <li...@lo...> - 2021-08-22 22:10:28
|
Hi Michael, Without FOP and no LXC containers, 1.0G RAM should be safe and not a worry. You could go lower, but you would have to monitor things more closely. Lonnie > On Aug 22, 2021, at 3:45 PM, Michael Knill <mic...@ip...> wrote: > > Hi Group > > I'm using VMware vCloud with one of my providers and have set up a Virtual Data Centre. I'm looking to set up a few Astlinux systems in this environment. > Although you can overcommit on CPU, you cannot on RAM and as this is fairly expensive, I'm wanting to go as low as I am comfortable on each Astlinux system. > > Just wondering what the maximum RAM usage you should ever see on an Astlinux system assuming no FOP is running? I have currently made it 1.5G but I think I can go lower than this. > > Regards > > Michael Knill > Managing Director > > D: +61 2 6189 1360 > P: +61 2 6140 4656 > E: mic...@ip... > W: ipcsolutions.com.au > > <image001.png> > Smarter Business Communications > > _______________________________________________ > Astlinux-users mailing list > Ast...@li... > https://lists.sourceforge.net/lists/listinfo/astlinux-users > > Donations to support AstLinux are graciously accepted via PayPal to pa...@kr.... |
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From: David K. <da...@ke...> - 2021-08-22 21:52:30
|
Can anyone recommend a good SIP trunk/pstn provider? I've been using Vitelity for years and had been happy with them, but recently long distance calls (well specifically calls from US to UK) have lost audio about 10-15 minutes into the call. Immediately calling back sometimes works, or sometimes it requires the other person to call me. I'm wondering if I should try a different provider. Any suggestions? Thanks David |
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From: Michael K. <mic...@ip...> - 2021-08-22 20:45:32
|
Hi Group I'm using VMware vCloud with one of my providers and have set up a Virtual Data Centre. I'm looking to set up a few Astlinux systems in this environment. Although you can overcommit on CPU, you cannot on RAM and as this is fairly expensive, I'm wanting to go as low as I am comfortable on each Astlinux system. Just wondering what the maximum RAM usage you should ever see on an Astlinux system assuming no FOP is running? I have currently made it 1.5G but I think I can go lower than this. Regards Michael Knill Managing Director D: +61 2 6189 1360 P: +61 2 6140 4656 E: mic...@ip...<mailto:mic...@ip...> W: ipcsolutions.com.au<https://ipcsolutions.com.au/> [IPC Solutions] Smarter Business Communications |
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From: Michael K. <mic...@ip...> - 2021-08-17 06:44:30
|
Thanks Lonnie.
Hmm that's a bit yucky. I suppose a third option is to use netset.
Regards
Michael Knill
On 17/8/21, 12:52 pm, "Lonnie Abelbeck" <li...@lo...> wrote:
Hi Michael,
You are not missing anything, there is no selective "Deny EXT->Local" as that is the default.
Two solutions come to mind ...
1) Recreate the "Pass EXT->Local" to multiple entries to not include what you don't want to allow.
2) Add a custom rule in /mnt/kd/arno-iptables-firewall/custom-rules to implement the desired "Deny EXT->Local".
-- untested example custom-rules --
deny_ext_local()
{
local proto="$1" host="$2" port="$3"
echo "[CUSTOM RULE] Deny EXT->Local for Proto: $proto, Host: $host, Port: $port"
iptables -A EXT_INPUT_CHAIN -s $host -p $proto --dport $port -j POST_INPUT_DROP_CHAIN
}
deny_ext_local udp 1.2.3.4 5060
deny_ext_local tcp 1.2.3.0/24 5061
--
(and test)
Lonnie
> On Aug 16, 2021, at 8:02 PM, Michael Knill <mic...@ip...> wrote:
>
> Yes.
>
> Regards
> Michael Knill
>
> On 17/8/21, 10:35 am, "Lonnie Abelbeck" <li...@lo...> wrote:
>
> Are you saying you added a "Pass EXT->Local" but now want to deny a subset of that ?
>
> Lonnie
>
>
>
>> On Aug 16, 2021, at 6:20 PM, Michael Knill <mic...@ip...> wrote:
>>
>> Hi Group
>>
>> Forgive my ignorance but just wondering how I do this. I want to block some addresses trying to register to the box from external.
>> PS its not SIP bots etc. Its known trusted addresses.
>>
>> Thanks
>> Regards
>>
>> Michael Knill
>> Managing Director
>>
>> D: +61 2 6189 1360
>> P: +61 2 6140 4656
>> E: mic...@ip...
>> W: ipcsolutions.com.au
>>
>> <image001.png>
>> Smarter Business Communications
>>
>> _______________________________________________
>> Astlinux-users mailing list
>> Ast...@li...
>> https://lists.sourceforge.net/lists/listinfo/astlinux-users
>>
>> Donations to support AstLinux are graciously accepted via PayPal to pa...@kr....
>
>
>
> _______________________________________________
> Astlinux-users mailing list
> Ast...@li...
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From: Lonnie A. <li...@lo...> - 2021-08-17 02:52:01
|
Hi Michael,
You are not missing anything, there is no selective "Deny EXT->Local" as that is the default.
Two solutions come to mind ...
1) Recreate the "Pass EXT->Local" to multiple entries to not include what you don't want to allow.
2) Add a custom rule in /mnt/kd/arno-iptables-firewall/custom-rules to implement the desired "Deny EXT->Local".
-- untested example custom-rules --
deny_ext_local()
{
local proto="$1" host="$2" port="$3"
echo "[CUSTOM RULE] Deny EXT->Local for Proto: $proto, Host: $host, Port: $port"
iptables -A EXT_INPUT_CHAIN -s $host -p $proto --dport $port -j POST_INPUT_DROP_CHAIN
}
deny_ext_local udp 1.2.3.4 5060
deny_ext_local tcp 1.2.3.0/24 5061
--
(and test)
Lonnie
> On Aug 16, 2021, at 8:02 PM, Michael Knill <mic...@ip...> wrote:
>
> Yes.
>
> Regards
> Michael Knill
>
> On 17/8/21, 10:35 am, "Lonnie Abelbeck" <li...@lo...> wrote:
>
> Are you saying you added a "Pass EXT->Local" but now want to deny a subset of that ?
>
> Lonnie
>
>
>
>> On Aug 16, 2021, at 6:20 PM, Michael Knill <mic...@ip...> wrote:
>>
>> Hi Group
>>
>> Forgive my ignorance but just wondering how I do this. I want to block some addresses trying to register to the box from external.
>> PS its not SIP bots etc. Its known trusted addresses.
>>
>> Thanks
>> Regards
>>
>> Michael Knill
>> Managing Director
>>
>> D: +61 2 6189 1360
>> P: +61 2 6140 4656
>> E: mic...@ip...
>> W: ipcsolutions.com.au
>>
>> <image001.png>
>> Smarter Business Communications
>>
>> _______________________________________________
>> Astlinux-users mailing list
>> Ast...@li...
>> https://lists.sourceforge.net/lists/listinfo/astlinux-users
>>
>> Donations to support AstLinux are graciously accepted via PayPal to pa...@kr....
>
>
>
> _______________________________________________
> Astlinux-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/astlinux-users
>
> Donations to support AstLinux are graciously accepted via PayPal to pa...@kr....
>
>
> _______________________________________________
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>
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From: Michael K. <mic...@ip...> - 2021-08-17 01:02:31
|
Yes.
Regards
Michael Knill
On 17/8/21, 10:35 am, "Lonnie Abelbeck" <li...@lo...> wrote:
Are you saying you added a "Pass EXT->Local" but now want to deny a subset of that ?
Lonnie
> On Aug 16, 2021, at 6:20 PM, Michael Knill <mic...@ip...> wrote:
>
> Hi Group
>
> Forgive my ignorance but just wondering how I do this. I want to block some addresses trying to register to the box from external.
> PS its not SIP bots etc. Its known trusted addresses.
>
> Thanks
> Regards
>
> Michael Knill
> Managing Director
>
> D: +61 2 6189 1360
> P: +61 2 6140 4656
> E: mic...@ip...
> W: ipcsolutions.com.au
>
> <image001.png>
> Smarter Business Communications
>
> _______________________________________________
> Astlinux-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/astlinux-users
>
> Donations to support AstLinux are graciously accepted via PayPal to pa...@kr....
_______________________________________________
Astlinux-users mailing list
Ast...@li...
https://lists.sourceforge.net/lists/listinfo/astlinux-users
Donations to support AstLinux are graciously accepted via PayPal to pa...@kr....
|
|
From: Lonnie A. <li...@lo...> - 2021-08-17 00:34:34
|
Are you saying you added a "Pass EXT->Local" but now want to deny a subset of that ? Lonnie > On Aug 16, 2021, at 6:20 PM, Michael Knill <mic...@ip...> wrote: > > Hi Group > > Forgive my ignorance but just wondering how I do this. I want to block some addresses trying to register to the box from external. > PS its not SIP bots etc. Its known trusted addresses. > > Thanks > Regards > > Michael Knill > Managing Director > > D: +61 2 6189 1360 > P: +61 2 6140 4656 > E: mic...@ip... > W: ipcsolutions.com.au > > <image001.png> > Smarter Business Communications > > _______________________________________________ > Astlinux-users mailing list > Ast...@li... > https://lists.sourceforge.net/lists/listinfo/astlinux-users > > Donations to support AstLinux are graciously accepted via PayPal to pa...@kr.... |
|
From: Michael K. <mic...@ip...> - 2021-08-16 23:20:24
|
Hi Group Forgive my ignorance but just wondering how I do this. I want to block some addresses trying to register to the box from external. PS its not SIP bots etc. Its known trusted addresses. Thanks Regards Michael Knill Managing Director D: +61 2 6189 1360 P: +61 2 6140 4656 E: mic...@ip...<mailto:mic...@ip...> W: ipcsolutions.com.au<https://ipcsolutions.com.au/> [IPC Solutions] Smarter Business Communications |
|
From: Michael K. <mic...@ip...> - 2021-08-14 23:01:49
|
Thanks Lonnie
Yes certainly using the Github page.
Regards
Michael Knill
On 15/8/21, 1:52 am, "Lonnie Abelbeck" <li...@lo...> wrote:
Hey Michael,
Looking forward to hearing how acme-dns works for you. AstLinux's acme-client (acme.sh) has a plugin for acme-dns, usage: --dns dns_acmedns
The acme-dns author "Joona Hoikkala" wrote an EFF article [1] "Securing the Automation of ACME DNS Challenge Validation"
BTW, I would use the acme-dns Github page [2] for info rather then the nethserver wiki article you referenced.
Lonnie
[1] https://www.eff.org/deeplinks/2018/02/technical-deep-dive-securing-automation-acme-dns-challenge-validation
[2] https://github.com/joohoi/acme-dns/
> On Aug 13, 2021, at 10:33 PM, Michael Knill <mic...@ip...> wrote:
>
> Actually decided that I will give acme-dns a try: https://wiki.nethserver.org/doku.php?id=userguide:let_s_encrypt_acme-dns
> Will report how I go.
>
> Regards
> Michael Knill
>
> From: Michael Knill <mic...@ip...>
> Reply to: AstLinux List <ast...@li...>
> Date: Saturday, 14 August 2021 at 12:29 pm
> To: AstLinux List <ast...@li...>
> Subject: [Astlinux-users] Securing DNS API Keys when using ACME
>
> Hi Group
>
> I'm looking to move away from Wildcard SSL and move back to ACME Lets Encrypt to ensure a unique cert for all our systems. The reason is that we have built our new Mobile Softphone solution which is heavily reliant heavily on TLS for provisioning and SIP.
>
> As such, I want to set this up but I am concerned that if one of our systems was compromised (we have quite a few now), this will allow an attacker to do bad stuff to our DNS (currently GoDaddy). I understand that some DNS providers may be able to restrict what you can do with the API but just wondering if anyone has any better ideas?
>
> Regards
>
> Michael Knill
> Managing Director
>
> D: +61 2 6189 1360
> P: +61 2 6140 4656
> E: mic...@ip...
> W: ipcsolutions.com.au
>
> <image001.png>
> Smarter Business Communications
>
> _______________________________________________
> Astlinux-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/astlinux-users
>
> Donations to support AstLinux are graciously accepted via PayPal to pa...@kr....
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