For logging SIP packets exchanged over TLS, enable "Listener" category at [Diagnostics]->[Debug Logs] page. For decoding/decrypting TLS and SRTP with Wireshark, the page below will help. https://www.zoiper.com/en/support/home/article/162/How%20to%20decode%20SIP%20over%20TLS%20with%20Wireshark%20and ...

Brekeke SIP Server's API doesn't provide Third-Party Call Control.

Do you want to do Third-Party Call Control?
https://docs.brekeke.com/pbx/third-part ... ontrol-pbx

> So, I need to use the right-most 11 digits, [Matching Patterns] $request = ^INVITE To = sip:.*(\d{11})@ You can test it at https://regex101.com/ Type sip:.*(\d{11})@ in "REGULAR EXPRESSION" field. and type sip:54872312015551234@xxx in "TEST STRING" field. The Group 1 will be 12015551234

Yes. you are right. You need the Group field.

If you want to make a Redundancy setup with Brekeke SIP Server, you need two unique Advanced Edition licenses and follow the instruction listed in the page below. https://docs.brekeke.com/sip/brekeke-sip-server-redundancy-setup Other redundancy solutions will not mirror SIP sessions between the ...

There are several ways to meet the requirement. - The easiest way is to use Brekeke PBX instead of Brekeke SIP Server because it has the feature to send SIP-REGISTER on behalf of SIP clients. - Another ways is to use Upper-Register at the Brekeke SIP Server to forward SIP client's SIP-REGISTER to ...

You can disable RTP-relay with $rtp = false setting in Deploy Patterns.

FYI:
https://docs.brekeke.com/sip/rtp

I think T2 is fine but I recommend T3.

Like this.

$session = failover sip:%1@<ip address two> sip:%1@<ip address three>

You can get the object of Session description from the method sdpContent.takeSessionDesc() . So the code will be like this. Sdp sdpContent = (Sdp)(Content.getContent(sippacket.content,Sdp.CONTENT_TYPE)) ; if ( sdpContent != null ) { SdpDescription sd = sdpContent.takeSessionDesc() ; if ( sd != null ...

have you read the section "7. Call Forwarding" in the document?

https://www.brekeke.com/doc/pbx/pbx_admin_advanced.pdf

Is it the purpose of DialPlan plugin? Note that DialPlan plugin is executed with each initial INVITE only. It means you can manipulate first SDP attached in an initial INVITE. If you want to manipulate all SDP including 200 OK's and re-INVITE's, you need to use Session Plugin instead of DialPlan ...

DialPlan can pick a content of SDP but it can not modify it.
Using Session plugin will be a solution but you need a coding.

Using Brekeke PBX is the easiest way.
http://www.brekeke.com/downloads/pbx.php

> 5 ## Created the jar file.
> /home/skb007/java> jar -cvf sni.jar sni.class

Since the package is "plugin", you need a folder named "plugin".

1) mkdir plugin

2) mv sni.* plugin

3) javac plugin/sni.java

4) jar -cvf sni.jar plugin/sni.class

> method=get_sip_call_id ex=java.lang.ClassNotFoundException: com.brekeke.plugin.sni The error message indicates that the plugin's class and/or method is not found. What Java package path did you specify for the class sni? Is it "package plugin"? If so, the package is "plugin". Put the line below ...

Which WebRTC client are you using?
Which web-browser are you using for WebRTC?

NDXBach,
Sorry. pls ignore my previous post. it was for another topic..

Can you see phone numbers in the [Registered Clients] page?

You don't have to use any DialPlan rules if a callee phone is registered in the SIP Server.
Let you disable or remove all DialPlan rules.


Anyway...
Deploy Patterns
To = sip:user

This is wrong definition.. It should be
To = sip:user@

Can you see phone numbers in the [Registered Clients] page?

You don't have to use any DialPlan rules if a callee phone is registered in the SIP Server.
Let you disable or remove all DialPlan rules.


Anyway...
Deploy Patterns
To = sip:user

This is wrong definition.. It should be
To = sip:user@

Try this. It rejects T.38 request with "603 Decline".

Matching Patterns:
$request = ^INVITE
$body("^m.+t38") = .+

Deploy Patterns:
$response = 603

Thank you!

1. Brekeke Product Name and Version:
Brekeke SIP Server 3.2

2. Java version:
Oracle Java 1.7

3. OS type and the version:
CentOS

4. UA (phone), gateway or other hardware/software involved:
NTT NGN

5. Your problem:
NTT NGN requires RFC3327. Does Brekeke support it?

1. Brekeke Product Name and Version:
Brekeke SIP Server 3.3.4.4

2. Java version:
1.7

3. OS type and the version:
Windows 2008

4. UA (phone), gateway or other hardware/software involved:

5. Your problem:

What kind of billing solution can I use with Brekeke SIP Server?