Brekeke PBX version 3.2 and later:
Brekeke PBX supports SRTP. The following steps show how to enable SRTP for the phone assigned to a Brekeke PBX user extension.
1. From Brekeke PBX Admintool > [Options] > [Phone Type] page, create a new phone type
2. At new phone type setting page,
– set [RTP Relay] “on”
– select proper setting at [SRTP] field
– set proper Audio codecs in [Codec Priority] field.
3. Save the new phone type and restart Brekeke PBX from admintool.
4. From a PBX user extension > [Phones] page, select the phone type created above at the [Type] field under the phone which needs to enable SRTP for the call. And save the changes.
5. Making calls from/to this user with SRTP feature enabled in phone type, Brekeke PBX will do the conversion between the users who donot support SRTP.
SRTP support can also enabled:
– ARS route valid for the calls through the route
– Brekeke PBX Admintool > [Options] > [Settings] page, valid for all the calls through the PBX system.
Brekeke PBX version 3.1.x and earlier:
Brekeke PBXdoes not support SRTP. However, you can pass SRTP strem through SIP Server by using the following Dial Plan to bypass SRTP.
[Matching Patterns:] $request = ^INVITE To = sip:9(.+)@ [Deploy Patterns:] To = sip:%1@
If a caller dials a number with the prefix “9”, the call will be forwarded to UA which supports SRTP directly. For example, if the UA’s number is “12345”, please dial “912345”