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

The RFC3261 defines it as
SIP-URI = "sip:" [ userinfo ] hostport

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.

FYI:
http://wiki.brekeke.com/wiki/Making-SIP ... r-Over-TLS

Hi loop,

Add the line below in the [Configuration]->[Advanced] page, and restart the Brekeke SIP Server. The developer fixed this bug already so upgrading the SIP Server to the latest version will also fix the issue.
http://www.brekeke.com/downloads/sip-server.php

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 interested in running both Brekeke SIP Server and Open VSwitch in the same VM guest? For production use, 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 = failover sip:32@192.168.10.222 &failover.timer.provisional = 5 With this rule, Brekeke SIP Server rings sip:30 ...

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-s ... ng-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 it causes a database lookup for each call. DialPlan table is loaded on memory so it doesn't cause any file ...

> 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, enable "video" at the JsSIP's GUI. > 2- How to configure the Dial Plan so a SIP client can call a WebSocket ...

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 = > ^100.6[4-9]|^100.[7-9][0-9]|^100.1[0-1][0-9]|^100.12[0-7] Because rev286.3 doesn't support net.internal-addr.pattern!! Let ...

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 internal API sets are available for OEM and tech partners.

Another way to fix the issue is .. Add this in the [Configuration]->[Advanced] page. 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 rule for REGISTER and you don't need "net.sip.fixed.addrport.uas" too.

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 honeypot rules to protect the server. http://wiki.brekeke.com/wiki/Block-List-Honeypot http://www.brekeke-sip.com ...

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-Aut ... SIP-Server

1. Brekeke Product Name and Version: Brekeke SIP Server ver3.2 Add following DialPlan rules in the [Dial Plan] -> [Preliminary] page. --------------------------------------- [Matching Patterns] $str.lowercase(User-Agent) = friendly-scanner|sundayddr [Deploy Patterns] $action = block [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, let you check your firewall settings. Make sure that UDP ports 10000-29999 should be opened at Window Firewall. http ...