Brekeke Forum - Search : Brekeke PBX, Brekeke SIP Server, Multi-Tenant Hosted PBX

> Route: <SIP:192.168.16.18:5060;lr>

The RFC3261 defines it as
SIP-URI = "sip:"

Try version 3 of Brekeke SIP Server because it accepts such ACK.

If you use a self-signed certificate as Client's certificate, you need to add it as a trusted certificate. Or set [Peer Certification Validation]=off at [Configuration] > [SIP] -> [TLS] page.
...

Hi loop,

Add the line below in the net.sip.packet.reader = false
The developer fixed this bug already so upgrading the SIP Server to the latest version will also fix the issue.
http://www.breke ...

We have some experiences with Open vSwitch (OVS) on CentOS and Ubuntu for Brekeke SIP Server.
In the evaluation phase, we use KVM and VirtualBOX for simulating the above environments.

Are you i ...

HTTP request's packet buffer size for WebSocket is 700 byte.

So resize the Cookie data to meet this limitation.

If you are using Brekeke SIP Server Advanced Edition, try DialPlan rule like this.
[Matching Patterns]
$request=^INVITE
To=sip:777@

[Deploy Patterns]
To = sip:30@192.168.10.222
$session = fa ...

If you use another web browser, can you export the list?

What kind of characters is it?

Are you planning to use Self-signed certificate?

Have you looked at this page?
http://wiki.brekeke.com/wiki/create-a-self-signed-certificate-using-OpenSSL

Are you using any special characters in user names?

It is an interesting.

I supposed Alias database meets the requirement.
You can add records with SQL from 3rd application, or CSV file.
http://wiki.brekeke.com/wiki/How-to-use-Alias-function
But ...

> 1- Will Brekeke Advanced support WebRTC Video call ?

Yes.
If both caller and callee (UAC and UAS) support WbRTC, you can use any media such as audio and video between clients.

For JsSIP, e ...

FYI:

Brekeke SIP Server version 3.4 and later support RFC6598 without any additional settings.
http://wiki.brekeke.com/wiki/BSS_History

tjtsaitsai,

Why are you using such a very old version?
The rev 286.3 was released 4 years ago and has some issues.

> the configuration by adding net.internal-addr.pattern =
> ^100. ...

Brekeke's developer team will add new GUI filed to tune the variable "net.internal-addr.pattern" in a future release.
If so, an administrator doesn't have to set it manually.

Some intern ...

Another way to fix the issue is ..

Add this in the net.internal-addr.pattern = ^100

Brekeke SIP Server will treat 100.x.x.x as an internal (private) IP address.

so you don't need any DialPlan ...

Are you using own SIP client developed on SDK?

Does the problem happen with both Xlite and your SIP client ?

> Because I would like to check if there is DDoS attack to my server which does not logging in error log.

But you can see them in the Blocked Log even if you are using ver3.2.

Add some hon ...

Did it happen during a phone call?
What SIP client are you using?

> It Seems that there is a problem with the connection authentication.

SIP Auth and RTP are not associated.

Did it happen during a phone call?
What SIP client and audio codec are you using?

> but Brekeke never send me an update.

Signup the eNews letter.
http://www.brekeke.com/company/enews.php

Added the Note.
http://wiki.brekeke.com/wiki/Set-up-Auto-Sync-Brekeke-SIP-Server

1. Brekeke Product Name and Version:
Brekeke SIP Server ver3.2

Add following DialPlan rules in the [Dial Plan] -> [Preliminary] page.

---------------------------------------
[Matching ...

bensly,
For your case, both SIP clients must enable RTP/AVPF instead of RTP/AVP.

How about the [Port mapping] setting in the [RTP] page?
It should be "source port" in your case.

If you try another SIP client such as Linphone..
Does the same issue happen?

Also, ...