It seems there is no problem because your deployment is simple.

Does the SIP Server connect to a SIP provider? Does the SIP Server connect to a VoIP gateway? Who is handling RTP packets? > I am also doing some logging in dial plan such as following. Is this going to cause some issues? Is it for statistics purpose? Since it adds new log line to the log, a little ...

> OS: Any recommendations for optimizations at OS level. Is it running on a physical machine? or virtual machine? > TOMCAT: Anything needs to be optimized at Tomcat level. Tomcat is used as web server. It doesn't affect call performance. > > Config->System->Java: Anything should go in the Java VM ...

It seems you are using version 2... it is too old (almost 10 years ago..)
Let you upgrade it to the latest version.

For outgoing calls, the definition should be "Contact" instead of "_Contact".

For response packet, you can use "_Contact".

You can manually Import/Export DialPlan file at [Dial Plan]->[Import/Export] page.


For automatic DialPlan backup, you may need to implement it with the management API or request Brekeke's team to add the function.

>or does this rule actually add the VIP to the OS on the secondary server? Yes. It supports both Windows and Linux. I recommend that you run Brekeke SIP Server with Administrator privilege. >Is it possible to monitor connection to CUCM, and have an action to failover if the CUCM is showing offline ...

&net.sip.replace.callid = true :Replace Call-ID, From-Tag and To-Tag &net.sip.replace.callid = false :Keep original Call-ID, From-Tag and To-Tag Try the following at Deploy Pattern. ------------------- $b2bua = true &net.sip.replace.callid = false ------------------- It keeps the original Call-ID ...

>We know the SIP Call-ID of the call which our application sends to BSS
> but we don't know the SIP Call-ID of the call which BSS sends to our
> provider.

Do you want to keep the original Call-ID what your application sends?

If so, remove "$b2bua=true" from the DialPlan.

If it is an urgent issue, I recommend you to contact support@brekeke.com . They will review logs and settings.

ok I got it.

Set [Configuration]->[RTP] page -> [RTP relay] = on

Is UA1 located behind a NAT (175.x.189.2)? Is 172.168.0.8 a local IP address of the UA1? If so, it should be a private IP address like 172.16.0.8... 172.168.x.x is not for a private IP address uses. If you want to treat 172.168.x.x as a private IP address, set ^172.168.* in [Internal IP address ...

Do you have an ARS route which can route a call to the specified phone number? In your example, it is 1234567.

> set ifsrc as media address(SIPSvr1's public address)

Why did you set $ifsrc?
Generally you don't have to use it because SIP server can put a suitable IP address in INVITE automatically.

Have you made a call without $ifsrc?
What kind of IP address are you using for the UA2's private network?

David,
Failover doesn't cover unknown host DNS error because the error happens before the SIP Server sends an INVITE to the first destination.
Failover plugin starts monitoring timeout or SIP response after the first INVITE was sent.

http://wiki.brekeke.com/wiki/sip-server ... top-resume

The evaluation license will stop running at random times ...

Which OS are you using at the client side? Have you captured RTP packets at the client side to confirm the packet dropping? Which audio and video codecs are you using? Do you have an enough network bandwidth to send/receive both audio and video streams? To confirm whether the packet dropping ...

I've checked it with the developer and confirmed that the BlockList function doesn't remove any users from the User Authentication database. These functions are running independently.

The only way to add/remove a user is from the Admintool.

Add DialPlan rule with the following Deploy Pattern in the [Preliminary].
This adds the parameter before the regular DialPlan rules are executed.

Deploy Patterns
To = <%{$getUri(To)}>;parameter=5

> 1.how can i change the SIP default listening port 5060. suppose i want to use 5080. It is the [Local Port] at the [Configuration]->[SIP] page. > 2.how can i change the TLS default listening port 5061(isn't it?). suppose i want to use 5081. If you set 5080 at the [Local Port], the TLS port will be ...

Why does SDP relate Call-ID? Call-ID is one of SIP header not in SDP.

For replacing Call-ID with a unique value, let you enable "B2B-UA mode" in [Configuration]->[SIP] page.

But there are no ways to specify "B" as the prefix of Call-ID.
If you write a plugin, it is possible.

Elidway,
Which version of Brekeke PBX are you using?

Call-ID is just unique string and it doesn't have any meanings.

Why does your application need to change it?
Does it manage both original and modified Call-ID?
Does it require "B" in the front of Call-ID?

Why do you need to change Call-ID?

Also "$getUri(Call-ID)" in your Matching Patterns will not work because Call-ID is not SIP-URI.