Announcements and off-site micellaneous destinations

Asterisk 13.17.0

We setup our FreePBX system to automatically forward calls received after hours off site to a 1-800 line via analogue POTS. And we created an announcement to tell callers that this is what is happening.

We used the Time Conditions and Time Groups applications to set up the after-hours checks. We used the System Recordings module to create the sound file that we wish played. We used the Miscellaneous Destination application to provide the target 1-800 number. We used the the Announcement application to play the sound file previously created and to subsequently go to the Misc. Dest., likewise previously defined. And we used the Time Conditions application to transferl to the after-hours announcement at the appropriate times.

The above simply provides the details of the setup process followed; both as an aide-memoire and to allow review for any defect or omission. The resulting setup appears to work as expected.

We have a few details that we would like to have smoothed out.

  1. Calls forwarded by this setup have an excruciatingly low volume for both parties. Is there any way to turn up the gain in both directions?

  2. There is a considerable delay in connecting off-site after the announcement has played and this is simply dead air. Is there any way to have the dialling start before the announcement has finished and to interrupt it if the line is picked up while playing?

  3. There is no ring tone from the time the announcement completes until the off-site line is answered. Is there a way to provide this?

  4. If the number is busy then the call is simply dropped. There is no busy signal returned to the in-call. Is there a way to configure our set up to report the busy condition; even if that involves another announcement?


I’m goin to guess (since it’s both ways and only through the POTS line) that the RXGAIN and TXGAIN need to be tweaked. Do not go nuts - the gain increases logarithmically.

There are couple of places to look at with these. The first would be to look at the trunk options and see when the audio actually starts. You may also want to look at the ‘R’ and ‘r’ options on the trunk so that the trunk (and not the remote device) generate ring. Another cool trick is to set everything up for FAX detection. This turns on the audio really early (since the system is them listening for a FAX machine).

There are a couple of approaches you could try. The simplest is to not use a misc-dest and instead set it up as an extension (for example) that has a voicemail as the destination (or ring busy and drop the call). There are actually lots of ways to do that in the system. Choosing one should be relatively straight forward. The one limiting factor for you will be the transfer of the call to the POTS line. Still, once you get the busy signal, you should be able to recover is you use one of the transfer mechanisms that have an “error destination”.

Thanks. I have not had a chance to try out your suggestions until today. In DAHDI Config - Global Settings I have increased the rx and tx gain by 0.1 and will see how much of an improvement that brings. I am not familiar with the insides of Asterisk and so, while I understand that the r option on the outgoing trunk will generate ring tones, per your recommendation, I cannot with assurance determine where this is actually set in FreePBX. I would appreciate be pointed to the correct module and setting location.

In the Global Settings there is a field called: Other Global Dahdi Settings: , but this has an implied syntax of option = value. Is this where the r option is set? Do I just place ‘r’ in the left hand side and leave the value empty? Or is this set somewhere else entirely?

Alternatively, I see this in each defined trunk listed on the Connectivity / Trunk page:

Asterisk Trunk Dial Options: Tt

Which I gather are the flags that permit the caller and the callee to each transfer the call should they wish.

So, do I modify this to: Ttr ?

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