>Either it will be a pre recorded audio file or user is talking on phone.

You can just set an empty personal message and make calls to the number. If you use a pre-recorded audio file, you can develop an application with JTAPI SDK.

The version of axis that is bundled with Brekeke PBX is 1.3.
I think any later version of axis works.

Anyway, I have never tried Tomcat 7 for Brekeke.
Please post the result if you try it.

Have you contacted the Aastra's support?
Echo problem usually depends on phones, not server.

I think you should not allow anyone you don't trust to log on your server. Don't you think so?

You can use audio file plug-in.

http://www.brekeke.com/download/download_pbx_doc_en.php
Developer's Guide - Audio File Plug-In

Good luck.

Brekeke has some development tools and APIs.

PAL would meet your requiremnt, too.

http://www.brekeke.com/download/download_pal.php
http://www.brekeke.com/download/download_pbx_doc_en.php

When you restart Brekeke SIP Server service, you need to kill all the Java processes befere starting the service.

Some type of sip phone can change the ringing sound by adding Alert-Info to INVITE request.
Do you know how can I do with CISCO phone?

Did you check the wiki?

http://wiki.brekeke.com/wiki/Brekeke_PB ... rosoft_OCS

What is it for?
What kind of project needs that function?

How if you disable your firewall?

Go to [Configuration]>[Advanced] and add a parameter like below.

net.sip.replacecontact=false

I have never tried this. But I notice this parameter in the version history.

>Is it possible to do multicast paging not unicast ?

I don't think there is a way to do multicast paging with brekeke.
Do you know what server products and SIP UA products support it?

PBX does not delete messages with Email notification. I think you can modify the configuration file. WEB-INF\work\pbx\en\tmpl\setting_tmpl.properties _config.137 Then change the number at options settings more than 900. This would help if you can make your plug-in. http://www.brekeke.com/download ...

Are you using third party database instead of hsqldb?
I think version 1.5 is too old and latest version 2 would work for you.

>All traffic from 190.x.x.162 must go to 192.168.2.1

You may want to configure like this.

[Maching patterns]
$addr=^190\..{1,3}\..{1,3}\.162

[Deploy patterns]
$target=192.168.2.1

What do you use SIPP for?
How did you configure SIPP?

The timeout works only when SIP packckets don't come before the timeout.
I think &net.sip.timeout.talking is not enough.

> -queries : _sip._udp.my_itsp_domain

This will not cause any problem.
Did you change any dialplan rules? Did you see any SIP REGISTR requests in your wireshark?

It does not support Speex, you can use iLBC.

My PBX had the same problem with Tomcat 5.5.
But It is working fine with Tomcat 6.6.

How do the REGISTRATION packets and the response packets look like?
Can you try capturing the packets and past them here?

I saw the same question at Japanese forum.

If you can read Japanese.
http://brekeke-sip.com/bbs_jp/viewtopic.php?t=34

<pbx-installdir>/WEB-INF/work/init.properties
com.brekeke.web.ParamUtil.MAXMEMORY_READ=10000000

Do you have superuser access?