If it is Responder SIP Server, there is a default DialPlan rule named "SIP_TAP". It handles SIP-TAP conversion.
Also refer to the Responder SIP Server's Reference Guide for more details.

Responder SIP Server converts SIP to TAP so Digione and Phillips don't have to support SIP.

Do you want to forward SIP message to a TAP transmitter through two Responder SIP Servers?

Like this?

Responder --(SIP)--> SIP Server 1 --(SIP)--> SIP Server 2 --(TAP/TCP)--< Digione SP --(TAP/serial)--> TAP Transmitter

Have you made users in the PBX?
https://docs.brekeke.com/pbx/brekeke-pbx-quick-start

If it is the bundled SIP Server in Brekeke PBX, you don't have to add or modify any DialPlan rules. The default DialPlan rules make all calls go to the PBX.
Instead of tuning DialPlan, let you use IVR in Brekeke PBX to meet the requirement.

FYI:
https://docs.brekeke.com/pbx/auto-attendant

Are you using Brekeke SIP Server aside from Brekeke PBX?
Or is it a SIP Server bundled in Brekeke PBX?

Do you open the RTP port range at Windows firewall?
In the default, the RTP port range is UDP 10000-10099.

The syntax is %{num}.
so %12 should be written as %{12}.

> - G:52.223.6.9 : domain: [<customer domain>.com] It means you have a setting for this domain under the [Domains] page. Because this domain's IP address is 52.223.6.9 and it is a public IP address, the SIP server puts it in Contact URI if 200 OK is sent to another public IP address. Do you have any ...

Are you sure no "52.223.6.9" at [Interface address 1] in [Configuration]->[System] page?

If you execute, "iflist" command at [Tools]->[Command] page, how is this IP address shown?

Does this IP address physically face to CUCM? or Load Balancer?

Do you have any other DialPlan rules which match an outbound call prior to the rule which executes the session plugin?

During a call, let you check the [DialPlan-Rules] filed in [Session Details] page from [Active Sessions].
If the field indicates another rule name, the plugin was not called.

Generally such information is available to partner companies and distributors.
But contact support@brekeke.com if you are interested in.

If your company has a support contract with Brekeke, contact support@brekeke.com to open a support ticket. We will look at Wireshark capture and Brekeke's log files to analyze the issue further.

> I didn't realize I had to put &net.sip.timeout.bye=1 in the PBX rules as well. It closes the canceled session immediately. so next call will not be blocked if the previous call was canceled. > So for example if the nurse call is ringing a "normal" call it will be To=sip6001@IPADDRESS > If an ...

Have you put "&net.sip.timeout.bye = 1" in the rule which forwards a call o the PBX? Also the blocking rule must be listed above the forwarding rule. Rule: Block the second INVITE [Matching Patterns] $request = ^INVITE To = sip:(6.+)@ $isTalking(From) = ^true [Deploy Patterns] $response = 486 &net ...

> So it changes the priority and sends to a different group of phones. That's why $isTalking(To) doesn't work because To-URI will be differ. > What is the sip timeout supposed to accomplish? If you add "&net.sip.timeout.bye = 1" in the Deploy Patterns, the SIP Server closes the Ringing session ...

Try this..

[Matching Patterns]
$request = ^INVITE
To = sip:(6.+)@
$isTalking( From ) = ^true

[Deploy Patterns]
$response = 486
&net.sip.timeout.bye = 1

It seems the original blocking rule like the following is better so far because the second INVITE has the same From-URI. [Matching Patterns] $request = ^INVITE To = sip:(6.+)@ $isTalking( From ) = ^true [Deploy Patterns] $response = 486 But it also blocks the next valid call too because the SIP ...

I see.If you have " $isTalking(From) " in that rule, any SIP INVITE which have same caller-ID (as From-URI) will be blocked. If you have " $isTalking(To) " in that rule, any SIP INVITE which have same callee-ID (as To-URI) will be blocked. I thought re-send INVITE has same From-URI and To-URI as the ...

Does the nurse call change the Caller-ID in From: header? I thought it changed Callee-ID in To: header. Can you try the DialPlan rule below. I added "$isTalking(To)". Matching Patterns $request = ^INVITE To = sip:(6.+)@ $isTalking(From) = ^true $isTalking(To) = ^true Deploy Patterns $response = 486

Such rejected calls are listed under [Logs]->[Error Logs] page.

> The only thing that changes is the first number in our "From sip: (#.+)

Does From-URI have the '#' character?
Do you need to catch an INVITE which has such a From-URI?

> Will this rule apply to any dial plans with a lower priority after it? Yes. but it only matches if "To = sip:(6.+)@" is fulfilled. > Also let's say the first number is 103, this rule will only block from 103 I assume? Yes. > Let's say a call from 105 is made, this one should still come through ...

Is the duplicated second call made while the first call is still ringing? If so, add the DialPlan rule below above the current rule. This rule returns "486 Busy Here" if the caller is on another SIP Session. Matching Patterns $request = ^INVITE To = sip:(6.+)@ $isTalking( From ) = ^true Deploy ...

First, there are several syntax errors in your DialPlan rule so it seems the DialPlan rule is never executed. Here is the modified DialPlan rule. Matching Patterns: $request = ^INVITE $registered = ^false To = sip:(6.+)@ From = sip:(1.+)\*(.+)\*(.+)@ Deploy patterns: To = sip:%1@IP ADDRESS From ...