Let you send the SIP-INVITE's Call-ID with APNS notification.
so the app can confirm whether it is a previously received call or not.
If it is a received call, the app should ignore that APNS notification.

> issues with the APNS arriving before the SIP invite.

Do you want to get a SIP-INVITE before the APNS?

My BSS Adv (v3.9.5.8. redundancy function test fail but I use BSS Adv (v3.14.3.5 trial version) redundancy function is ok.(ps: same environment except BSS) Because there are many changes including bug fixes from v3.9.5. https://docs.brekeke.com/sip/sip-history Q1 :If I want to upgrade my Brekeke ...

It seems that the Brekeke SIP Server is running without any issues.

Can you access to the Brekeke SIP Server's GUI page?
If so, Brekeke SIP Server is running with Amazon Corretto.

Because Brekeke SIP Server supports Java 11, there will be no problems.

> // only the history from the SIP phone from 172.27.0.123. no other records. 172.27.0.123 is the Server's IP address. Do you run a SIP phone app on the server's localhost? According to your old post, SIP-REGISTER was sent from 172.27.1.3. Can you still send a SIP-REGISTER from 172.27.1.3? If so ...

Can you send a ping to 172.27.0.123 from other phones network environments? Did you set "172.27.0.123" as the SIP Registrar's IP address at other phones? Can you see SIP-REGISTER sent from other phones at SIP Server's [Dial Plan]->[History] page? Can you see other phones IP address at [Block List ...

The rule "outbound call" has the issue because it doesn't have any Matching pattern conditions. (it means this rule catches REGISTER request too..) Generally, you don't need any DialPlan rules for outbound calls. So can you remove the "outbound call" rule from the DialPlan page, and push [Apply ...

Can you see 172.27.0.123 at [interface] field in [Status]->[SIP Server Status] page?

Did you edit the DialPlan file "dialplan.tbl" directly?

Is the SIP Server running on 172.27.0.123?
Can you see 172.27.0.123 at [interface] field in [Status]->[SIP Server Status] page?

Did you get "not found" error just after the SIP phone sent a SIP-REGISTER request?

Are you using any DialPlan rules at your Brekeke SIP Server?
Are you using SIP authentication at your Brekeke SIP Server for SIP-REGISTER? (it is "on" in the default.)

"^Invite" should be "^INVITE" so the Matching pattern will be like this

Matching Patterns:
$request = ^INVITE
From = sip:12223334444

Matching Pattern is depending on your routing needs.

Simply it can be written as
Matching Pattern
$request = ^INVITE

That's why you need to specify the IP address of the upstream service provider at the redirect server in STIR/SHAKEN service.

You may need to request a license transfer if it was activated with Offline activation method.
https://docs.brekeke.com/lic/license-transfer-request

Otherwise, re-activate the license at new location.
https://docs.brekeke.com/lic/how-to-mov ... ion-method

If you want to make call from a WebRTC client such as JsSIP, a callee client must support ICE and DTLS. So you can make call from JsSIP to another JsSIP without any issues. But most of non-WebRTC SIP clients such as Cisco can not accept a call sent from WebRTC. Please refer to the pages below ...

Have you tried 3.12.1.5 with AdoptOpenJDK 11? If it was 3.9.1.3, the installation will fail because that version didn't support AdoptOpenJDK. If you want to try AdoptOpenJDK 11, please use a version 3.12.x.x The latest version is 3.12.2.2. https://docs.brekeke.com/sip/sip-history https://www.brekeke ...

I tested your DialPlan rule in my BSS 3.8.5.2 and I confirmed that the From-URI is NOT replaced without tuning the rule.
It means 3.8.6.4 and 3.8.5.2 are same in this case.

If you paste all of your DialPlan rules you set at 3.8.5.2, I can look at them.

In your first post, why did you set "$addr=192.168.2.10" in the Matching Patterns?

Because 192.168.2.10 is one of BSS's interface addresses, $addr must be 192.168.2.5, UAC's IP address.
$addr = 192.168.2.5

If you have $addr=192.168.2.10, the DialPlan rule doesn't work.

$time("HH:mm:ss.SSSS")

I recommend that you read the admin guide.

There are several conditions.
Is the UAC registered in the Brekeke SIP Server?
If so, Brekeke SIP Server replaces From-URI automatically.

Put this in Deploy Patterns

Deploy Patterns
From/uri = sip:%{$getSIPuser(From)}@192.168.10.11

Sorry.. because it depends on the version of Brekeke SIP Server.

Try $replaceuri = true

Put "$replaceuri.from = true" in the Deploy Patterns.

> Yes, I've checked "Authentication/Radius" at the [Debug Logs] page. Can you find "auth: NG" message in sv.xxxx.log? > Is it possible that the default log level doesn't show it? The default log level is 0 and it doesn't show detailed information. >How do I increase the level of logging of the BSS ...