For writing log from the RTP-relay, put the following line in the [Configuration]->[Advanced] page. ---------------------------- net.rtp.loglevel.file=255 ---------------------------- After you reproduce the issue, open the log file. C:\Program Files\Brekeke\sip\webapps\sip\WEB-INF\work\sv\log\2014 ...

How about audio quality?

What kind of internet connection does your customer use?
Does it provide enough bandwidth?

According to SDP, there are 3 media streams (1 audio + 2 video).
so.. the network bandwidth should be wide enough.

Glad to know!

To apply new JAR file, you need to reboot the OS.

The ver 2 doesn't have such a configuration.
The ver 3's RTP Relay runs as OS's native code.

The database structures are same in both v2 and v3. So you can refer the above PDF. For using MySQL.. have you copied MySQL's JDBC driver like the following? mysql-connector-java-5.1.31-bin.jar You need to download it from the MySQL's site. https://dev.mysql.com/downloads/connector/j/5.1.html And ...

If you use $log in Deploy Pattern, you can write own log message in sv.xxx.log file.

Find "$log" in the http://www.brekeke.com/doc/sip/sip_admin_v3.pdf

For example...
[Deploy Patterns]
$response = 503
$log = WARNING! Call limit is reached!! Return 503 to %{$addr}.

If you use a database for Register table, you can remove certain user with SQL.
http://www.brekeke.com/doc/sip/3rd_party_database.pdf

Add the following in the [Configuration]->[Advanced] page once, and restart the SIP Server. Now you can catch REFER with DialPlan even if it is sent in existing dialog.
Let you write DialPlan rule to catch REFER and modify it as you want.

Which video product are you using?

We don't know what dialed numbers are "incoming" or "outgoing" in your case.
They are depending on how you deploy voip service.

Use the To-URI. It is the dialed number.

As Hope said, there is an automatic protection which blocks an IP address after few login failed.

Also, Brekeke has the web API which allows you to make own user-login page.
With this API, you can design and develop secure user-login page.
Let you contact Brekeke for more details.

> From=sip:(.+)@(.+)

It must be:
From = <sip:(.+)@(.+)>
Or,
$getUri( From ) = sip:(.+)@(.+)

Do you set the correct SIP Server's IP address as the register server at the setting of X-lite?

Is it FQDN? or is it IP address?

Brian,
Which version of CS1000/SIP trunk are you using?
Are you using Avaya SES too?

Nortel CS1000 was working well without issue but it seems Avaya did some modifications..

Try the settings below:

In the [Configuration]-[SDP] page,
[RTP relay] = on
[Port mapping]= sdp
[Send UA's remote address] = no

Do you want to receive an e-mail notification if a callee doesn't pick a call?

I know the following Brekeke PBX's API allows you to create such a feature.
http://www.brekeke-sip.com/download/bpb ... _db_en.pdf

Are you talking about Brekeke PBX's feature?

In the SIP Server, go to the [Server Status] page.
There is the [work-directory] field.


You can find sv.xxx.log under the following folder.
<work-directory>\webapps\proxy\WEB-INF\work\sv\log\2012\08\

Vinnie, > Client (Cisco 7921) is reg as 7000 > sip:7000@192.168.27.202:5060 Do you mean the CISCO 7921 makes a registration to the Brekeke SIP Server? > I have noticed there that on the requester settings it comes up as > 192.168.27.125:1949 not :5060 like the other 2 phones. You don't have to worry ...

Hi JasonIsrael,

It seems you are using Brekeke PBX so you need to use PBX's ARS for these requirements.


If you are Brekeke's dealer, I think you can contact Brekeke's support team directly.

Set "net.sip.fixed.addrport.uas = true" in the SIP Server's [Config] -> [Advanced] page, and reboot the SIP server.

With this setting, the ACK packet must be sent to the callee (UAS).

I doubt that you met the requirement.

Can you make a call through the SystemB when SystemA is down?

Add the following in the Deploy Pattern.
---
&net.registrar.useoriginal = true
---

Do you want to allow a phone to make a call even if the System A is down? To prevent an unregistration, add the following DialPlan rule before the current rule. ------------------------ Matching Pattern $request = ^REGISTER Expires = ^0$ Deploy Pattern $auth = false $action = register &net.registrar ...

--------------------------
Matching Patterns

$request = ^INVITE
From = sip:(6[0-9]{4}).+@
To = sip:9(.+)@
$regaddr( "%11" ) = (.+)


--------------------------
Deploys Patterns

To = sip:%2@%3
-------------------------

It seem you did a pretty good job.

Try this.
--------------------------
Matching Patterns

$request = ^INVITE
To = sip:100(.+)@
$regaddr( "100" ) = (.+)


--------------------------
Deploys Patterns

To = sip:%1@%2
-------------------------