How to Make Specific Calls "unanswerable" but Stil
Moderator: Brekeke Support Team
How to Make Specific Calls "unanswerable" but Stil
1. Brekeke Product Name and Version: Responder SIP Server 3.14.0.7/557
2. Java version:
Eclipse Adoptium 11.0.17
3. OS type and the version:
Windows Server 2019
4. UA (phone), gateway or other hardware/software involved:
Whistle iOS Softclient
5. Your problem:
Need a way to prevent a phones from accepting/establishing audio specific INVITES. I can match on the types of calls but have not found a deploy pattern that prevents it from "answering" the call
2. Java version:
Eclipse Adoptium 11.0.17
3. OS type and the version:
Windows Server 2019
4. UA (phone), gateway or other hardware/software involved:
Whistle iOS Softclient
5. Your problem:
Need a way to prevent a phones from accepting/establishing audio specific INVITES. I can match on the types of calls but have not found a deploy pattern that prevents it from "answering" the call
How about.. rejecting a call with a SIP response such as "404 Not Found" with $response.
https://docs.brekeke.com/sip/response
[deploy pattern]
$response = 404
https://docs.brekeke.com/sip/response
[deploy pattern]
$response = 404
I suppose, we send a specific call type to multiple phones, if a phone declines the call their phone stops ringing while the others continue. If someone accepts the call and then hangs up we start ringing all the handsets again. That's why I'm trying to suppress the ability to accept the call. Does that make sense? I know it's a tad convoluted
I suspected as much, my customer is not entertaining the upgrade to the PBX product and was trying to configure the SIP Server to do it. I was able to do it with a softclient by providing a codec payload the client couldn't use but when doing this with a mobileheartbeat device it won't ring the phone with an unusable codec
The INVITES originate from the Nurse Call system which is generating a "call" based on activity initiated in the Nurse Call System. The event is an urgent level call which is why after an accept we send a fresh invite to all of the recipients of the call. Since we add the "call type" in the calldisplay portion of the From field I can reliably match on that value but I don't want to match on the INVITE, I want to match on the OK the phone replies with for that call.
Feicstur,
Is it for terminating a call that the UAS accepted with "200 OK"?
Since these SIP response codes are "Final response" according to RFC3261, the UAC (caller SIP client) might ignore 404/603 because it already received "200 OK".
Why do you want to terminate a call with 404 or 603 even after the call was accepted with "200 OK"?
Is it for terminating a call that the UAS accepted with "200 OK"?
Since these SIP response codes are "Final response" according to RFC3261, the UAC (caller SIP client) might ignore 404/603 because it already received "200 OK".
Why do you want to terminate a call with 404 or 603 even after the call was accepted with "200 OK"?
My idea is to set the talking timer 1 ms.
Put the line below in the Deploy Patterns of the rule.
If so, the call will be terminated just after 1ms from "200 OK".
The default value is 259200000 [ms] (=3 days) which you can tune at [Talking Timeout (ms)] in the [Configuration]->[SIP] page.
Put the line below in the Deploy Patterns of the rule.
Code: Select all
&net.sip.timeout.talking = 1
The default value is 259200000 [ms] (=3 days) which you can tune at [Talking Timeout (ms)] in the [Configuration]->[SIP] page.
It depends on existing DialPlan rules.
Did you put "&net.sip.timeout.talking=1" in the current rule? or new rule?
If it is new rule, you might need to add $continue=true in Deploy Patterns too.
[Deploy Patterns]
&net.sip.timeout.talking = 1
$continue = true
If you still get "404", other DialPlan rules may be influencing.
Did you put "&net.sip.timeout.talking=1" in the current rule? or new rule?
If it is new rule, you might need to add $continue=true in Deploy Patterns too.
[Deploy Patterns]
&net.sip.timeout.talking = 1
$continue = true
If you still get "404", other DialPlan rules may be influencing.
Have you push the [Apply Rules] button after you modify any DialPlan rules?
What kind of SIP entity are you using as the call destination?
Is it registered in the SIP Server or did you specify the destination's SIP-URI or IP address with any DialPlan rule?
Does the destination phone ring if you disable the DialPlan rule which has "&net.sip.timeout.talking"?
What kind of SIP entity are you using as the call destination?
Is it registered in the SIP Server or did you specify the destination's SIP-URI or IP address with any DialPlan rule?
Does the destination phone ring if you disable the DialPlan rule which has "&net.sip.timeout.talking"?
> Disabling the dial plan rule rings the phone
Have you put "&net.sip.timeout.talking=1" and "$continue=true" in the Deploy Pattern of the rule?
> Disabling the dial plan rule rings the phone
Check the rule names which applied the call.
Go to [Logs]->[Session Logs] page, click the date and find the rule name at [Rule Name] filed.
Are there any rule names there other than "registered=" ?
Have you put "&net.sip.timeout.talking=1" and "$continue=true" in the Deploy Pattern of the rule?
> Disabling the dial plan rule rings the phone
Check the rule names which applied the call.
Go to [Logs]->[Session Logs] page, click the date and find the rule name at [Rule Name] filed.
Are there any rule names there other than "registered=" ?