Siprecorder Settings

SIP Settings


Use the Unique Call ID to keep track of the calls that are recorded.
The FROM and TO fields will be cleared in the tracking code. Multiple calls can with the same Unique ID is then combined into one recording. USE WITH CAUTION.


Use the FROM + TO + CALLID to keep track of calls.
This can cause multiple calls to be recorded if the multiple Sources and Destinations are involved and the capture is seeing all the traffic in the mirroring.



The 3CX has a Click to dial facility that allows a soft phone to control a Physical phone. the issue on the recording side is that the calls show as being initialed from the PBX side to the Physical phone. So Outgoing calls show as Incoming calls.
This setting looks for the MakeCall facility in the SIP data and changes the call direction to the opposite direction.


The Avaya PABX has another method to update phone information via HTTP requests and this facility allows the Extension to be retrieved from these requests. If this option is disabled that some phones may not be updating their extensions.


The Siprecorder will clear auto detected extension numbers that are associated with IPs after 24 hours. This setting can change the number of hours before it is cleared. This is useful to use with the AVAYAHTTP option, in case people do not login over weekends and the phones loose their extensions.


With the excluded section you can add extensions that you do not want to record, when you have RECORDALL=1. This allows for better control where you want to disable some extensions or Main numbers that you want to exclude.


This setting allows SIP recordings to stop recording after this amount of seconds if no further RTP is received. This is useful in sites where there are long 4 hour recordings being made that doesn’t stop because the stop message was not received.



Added support for wildcards and CIDR for extension values. Use something like:


Fixed spelling error in extrainfo, introduced in I think.

Added CIDR support for the white-list and black-list.
You can use in the Black and White lists.


Attempted Avaya crash fix.

SIP ports are now configurable - previously they were hardcoded to 5060, 5061, 18060, and 35060


Added option to record only if both RTP ports for a call must be present in a packet to be assigned to the call. This is necessary for shared PABXs/border controller that can re-use a local port if the IP sending to that port is different.>


It is disabled by default (BOTHPORTS=0), which is the old default behavior.
Usernames that are the same as the extension are now prefixed with “EXT” to allow Sorter to use mapping username instead of having the extension appear there.
Added configurable option to use Remote-Party-ID to get the username.


Default is 0, which is the old behavior.