I think that is only for Voicemail, Brekeke PBX can not send recorded files by e-mail.

Andrea

I'm not sure but maybe you can set the field like the one in Dial Plan Matching Patterns.
Such as ^10.*|^11.*


Andrea

I don't think they support SIP & RTP encryption.

Some phones support RTP encryption.
Brekeke SIP Server can relay encrypted RTP.
So it will work fine with Brekeke SIP Server.

Brekeke PBX doesn't handle encrypted RTP.

Andrea

Your question is not related to Brekeke JTAPI SDK. Just include those downloaded java files into the build path of your development environment and implement JTAPI, then create your own application. There are detailed documentations for the JTAPI specification in the same sun site. I recommend you ...

Which header do you want to add the info (caller ID & DNIS) to?
Can't you just forward the call to external provider?
Then I think From header should include the caller ID and To header includes DNIS.

Andrea

Hi, You should not set a user number in action. $action = 603 means the error response 603 (Decline) will be returned to the caller. Example (to ban the user 9001 to call international call): Matching Patterns: $request = ^INVITE To = sip:9(.{10,})@ From = sip:9001@ Deploy Patterns: $action = 603 ...

Hi,

Why do you need to use the Dataflex gateway for the communications over the internet? Do you want to make a VPN and let SIP communications go through the VPN?
I'm just curious.
I don't think you need two SIP Servers. Just one SIP Server is enough.

Thanks,
Andrea

Hi, How about this? Matching Patterns $request=^INVITE to=sip:(90.+)@ $time=^..:[0-2] (change 30 min) From = sip:(.*)@ Deploy Patterns From = sip:user2%2@<pbx_IP_address> target = localhost:15060 -------------------------------------------------------------- Matching Patterns $request=^INVITE to=sip ...

Hi,

Check Admintool's [Sessions] menu.
While you are talking, is the session status "Talking"?

Andrea

Hi,

This may be useful.
http://www.brekeke-sip.com/wiki/index.p ... i=195ce9d6


Andrea

I think Presence & Message feature of WM is not supported by Brekeke. But I think you can make a call each other. Try

Andrea

Hi,
There is RTP Session Timeout setting in both PBX and SIP Server.
If RTP go through the server and no RTP packet received for the period of Timeout, the call should be disconnected.
Check those settings.

Andrea

It may be a codec.
Check which codec is set in the phone.
Try set PCMU (G.711 ulaw) for all phones.

Andrea

Hi Han,

I got the same error message with sample4.
But all each user could hear other users voices with no problem.
I think the sample is no problem.

What phone did you use for user3?
Is microphone enabled?

Andrea

Hi,
I think the link http://www.brekeke-sip.com/bbs/viewtopic.php?t=1096 is helpful

Andrea

>3) I disabled the "To PBX" rule but didn't work.

What error response did you get?

Andrea

JSTsai,

Try adding current directory "." in your class path.

Andrea

Hi,

"java.lang.NoClassDefFoundError"
It's still classpath problem.

Run with the following command:
java -classpath <specify jar path>;<jar path>;<jar path> CalloutPlay


Andrea

You need to include all jar files that Brekeke provided in classpath.

Cheers,
Andrea

I mean the port that JTAPI uses, not SIP Server.
You set the port in provider.properties (mysipport).

Andrea

If you try to open 2 or more JTAPI applications at the same time on the machine or if you are running some other machine which is using the same port as your application, the error occurs.

Andrea

Hi gordsh,

It is 8 k, 1 channel, 8 bit "U-LAW" encoding.
You can use SoX http://sox.sourceforge.net/ to convert from other format of sound file to .ul.

Andrea

If you install Brekeke JTAPI, userdata file should be created in the directory. You need to have userdata and license files in the directory to run your application.

Andrea

Usually, PSTN gateway is hardware.
See http://www.brekeke.com/support/support_ ... ateway.php
for example.

Or, you can use some telephony provider such as vonage to call out to GSM phone. See http://www.brekeke.com/support/support_ ... y_itsp.php .

Andrea

Yes, you will need SIP Server and PSTN gateway.
See the datasheet http://www.brekeke.com/download/downloa ... doc_en.php .


Andrea