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

For writing log from the RTP-relay, put the following line in the close: total.pkt=

It is the number of total received RTP packets per media stream.
If it is less than the actual number, the re ...

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 ...

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.j ...

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 ...

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 &net.sip.outside.dialog.pattern = ^REFER|^NOTIFY

Now you can catch REFER with DialPlan even if it is sent in existing dialog.
Let you write DialPlan rule to catch REF ...

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 ...

> 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/bpbx ...

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\ ...

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 ther ...

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.
------------------------
Matchin ...

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

$request = ^INVITE
From = sip:(6Deploys Patterns

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

It seem you did a pretty good job.

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

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

----------------------- ...