Call recording on non recorded extension

Hi All,

Sorry if this is not the right please for this topic but you are my last hope as some Vendors that use Asterisk with their own hardware/software refuse to help or implement some of the functionalities!
The issue I am facing is regarding the Phone recording functionality!
The intended setup is :

  • outside call to CS department extension (on call pick-up phone recording starts)
  • CS transfers call to non recorded extension (on call transfer to non recorded line recording stops)
  • If call is transferred back to the CS extension than call recording restarts on the same file or creates a new record
    The current situation is that all calls get recorded regardless of call transfer or if the call is transferred to a non recorded extension! If the call is received on the non-recorded extension then no call recording takes place, which is the desired outcome.
    The Vendor I am talking about is Yeastar and the Asterisk Version is : Asterisk 13.7.0 . Please let me know what other info I can provide and if this is even possible to do from Asterisk Config !

PS: I have it setup so that no local LAN calls get recorded



I don’t know enough about the call recording features to give a definitive answer without doing more research than is justified by a peer support forum response.

However, I think you have conflicting requirements. The selective recording suggests you want to tie the recording to the B side channels, but the same file requirement requires associating it with the A side. Calls do now get an identity associated with the oldest channel associated with the call, which you just might be able to use.

Also, anything related to transfers depends on the exact way in which the transfer was initiated, although in many case, the call to the transferee is set up before anything knows that that here is going to be a transfer.

Hi David.

Thanks for your reply. What I am trying to achieve first is to stop call recording on call transfer!
Because I’m a newbie when it comes to PBX’s, can anyone point me in the right direction?
Now, I know I can pause the call recording and start it again with the feature code *00 , which helps me a little bit, in the console I see :
– === PAUSE MIXMON [DAHDI/i2/xxxxxxxx-b3] === .
– === UNPAUSE MIXMON [DAHDI/i2/xxxxxxx-b3] === .
Is there any way I can send this command (*00)to the PBX as soon as the extension picks up so the recording stops??
Can I

*00 only does this because that’s what features.conf tells it to do. The default one doesn’t even to that, so you must have added that yourself, or used third party stuff that does.

You can’t send features codes to extensions, only to devices and channels. You can only send internally sourced ones to channels if there is a local channel in the call. In any case, sending internally, to control other parts of Asterisk is always a kludge.

Unfortunately the processing on the answer for a transfer is quite complicated, and differs depending on how you request the transfer. I think there is a good chance that any subroutines run on answer are run before the A party is connected, and therefore on a channel that does not have MixMonitor running. True blind transfers may be better behaved, in which case you could probably pause before the Dial. However some SIP phones use attended transfer logic even when the user things they are doing a blind transfer.

That’s why I said you need to tell us exactly how you do the transfers.

Blind transfers would be the most common case. I expected asterisk to be able to pause a recording in any situation if the extension is set as NON-Recorded by default. Just as the monitor function, you can enable/disable on a per extension basis and have granular control over which monitoring function you need.

You are still confusing extensions and devices.

Monitoring is associated with neither is connected with channels. The handling of monitoring is procedural, whereas you seem to want a non-procedural behaviour.

There are three ways of doing blind transfers, at least:

  1. features.conf


  3. SIP INVITE followed by an automatic REFER/Replaces.

Case (3) has particular problems doing what you want, because the transferee call is set up before Asterisk knows that it is part of a transfer. Some SIP phones use this method.