Mine is like this.
[Matching Patterns]
$request = ^INVITE
$getSIPuser(From) = ^(\S+\*\S+)\*\S+

[Deploy Patterns]
From/uri/user = %1
$continue = true

If there are no INVITE sent from the CISCO to the Brekeke, it will be an issue at Cisco's settings.

> at this point i am never seeing an INVITE from cisco.

Are there any packets other than SIP-INVITE?

Have you looked at this Configuration Note?
https://brekeke.com/doc/sip/jeron_conf_ ... _bssv3.pdf

> $addr = ^xxx.xxx.101.xxx$|^xxx.xxx.102.xxx$ It works. Also it might be written as $addr = ^xxx.xxx.10[12].xxx$ If you allow the range between xxx.xxx.101.xxx and xxx.xxx.102.xxx, you can write $addrRange( "xxx.xxx.101.xxx", "xxx.xxx.102.xxx" ) = ^true FYI: https://docs.brekeke.com/sip/addr https ...

The client's SIP port should be tuned at the setting of the client.
So please refer to the phone's document.

But.. I don't think using the SIP port 5060 is the cause of the issue.
I recommend that you upgrade phones firmware.

Set the variable "net.sip.sesstimer.expires-min" in Brekeke PBX's [Options]->[Advanced] page.

The default value is 90.
If you want to tune it to 1800, put the line below.
net.sip.sesstimer.expires-min = 1800

Do you mean that the T20P (ext 116) doesn't have the issue but the T23G (ext 132) has the issue?

Does the issue happen only during the night?
Is the issue solved temporarily if you restart the Brekeke PBX Server?

Let you try to update both Java and Brekeke PBX because they are old specially Java 7..

If you want to pass SIP-OPTIONS to sip:200@x.x.x.x, try the DialPlan below. [Matching Patterns] $request = ^OPTIONS [Deploy Patterns] To = sip:200@x.x.x.x If you want to return "200 OK" to SIP-OPTIONS, try the DialPlan below. [Matching Patterns] $request = ^OPTIONS [Deploy Patterns] $action = 200

If you have a concern about logging's impact, I recommend that you enable the SIP session logging at only certain SIP calls with DialPlan. Refer to the Logging SIP packets through a certain dial plan in the wiki topic below. https://docs.brekeke.com/sip/how-to-log-sip-packets Adding " &net.sip ...

> 172.27.0.123 is the Server's IP address. Do you run a SIP phone app on the server's localhost? // yes, only this soft phone can be registered. It seems that your OS blocks SIP packets sent from non-localhost. Usually this issue is caused by a firewall. > Can i install it to other server or linux ...

> When a 3xx redirect occurs, is there any indication of that in the Brekeke logs? 3xx SIP response packet will be shown in sv.xxxx.log file. > Is there anywhere in the Brekeke logs to see the failover occurring? "failover" log line will be shown in sv.xxxx.log file. > What I find in the logs is ...

> Tomorrow I'm hopeful to be able to test at least partly with using &failover.pattern.response = 3..

Do you want to let the Brekeke SIP Server handle 3xx redirection responses without forwarding them back to the call originator?

> Would there be any difference between setting &failover.timer.inviting = 10 explicitly versus just not setting it at all (thus using the default)? They are same because 10 is the default value. > Could I set &failover.pattern.response = ^$ to test for a null string? Would that effectively ignore ...

Let you run the installer with Administrator account.

Hi ajlindy,

Have you looked at the following wiki topic "How to update web server (Apache Tomcat)" ?

https://docs.brekeke.com/sip/update-web-server

> Response of BSS to second INVITE
> ===============================
> INVITE sip:85.162.2.37:5060;lr;transport=udp SIP/2.0

It is not a response from the BSS. It indicates an INVITE request packet proxied by BSS and sent to the UAS (85.162.2.37).

Have you looked at the BSS's log?

According to these packets, From's tag and Via's branch parameter are different between the First INVITE and the Second INVITE, so it seems the Second INVITE is not re-transmitted packet of the First-INVITE. But the CSeq in the Second INVITE is still "CSeq: 1 INVITE". It is the problem. It should be ...

Add the setting below in the Deploy Pattern. [Deploy Pattern] &failover.redirection = false If you want to disable the redirection in the entire SIP server, add the setting below in [Configuration]->[Advanced] page. failover.redirection = false FYI: https://docs.brekeke.com/sip/failover-with-the ...

For removing a certain SIP header, put the header name with no value in Deploy Patterns.

For example
[Deploy Patterns]
P-Preferred-Identity =

&failover.timer.provisional is the timer applied after any provisional response is received. If you need a timer just after an INVITE is sent, use &failover.timer.inviting. The default value is 10 (sec) https://docs.brekeke.com/sip/failover-with-the-dial-plan [Matching Patterns] $request = ^INVITE ...

Why did you put "&failover.timer.provisional = 35"?
If you don't want to make a failover after any provisional response (1xx)
is received, let you set a big number there such as 86400 (=1 day)

&failover.timer.provisional = 86400

Your rule will be like this if you use Alias
[Matching Patterns]
$request = ^INVITE
$alias.lookup($addr) = .+
To = sip:(.+)@
From = sip:(.+)@

[Deploy Patterns]
To = %1@4.4.4.4
From = %2@5.5.5.5

At [Aliases] setting, add IP address as each record, or import a CSV file which lists IP addresses.

> If we were to tell Brekeke to ALLOW address from two or three specific IP addresses, would it by default BLOCK everything else? No. An IP address marked "Allow" should not be blocked even if the BlockList detects malicious activities from the IP address. If an IP address is marked "Block", the ...

Execute the command below at [Tools]->[Command] page. log sys -i Does the sizelimit's value indicate 600MB or more? If not, execute the command like this. log sys -l 600000000 And check the sizelimit's value again. log sys -i If the above solution doesn't work, execute the command below. log sys -n