Volume function is in which module?

I have Asterisk 16 installed, but it was a migration from Asterisk 11.3. In my script, I used a function called “Volume” but in 16 it says “function not registered”, so which module I need to load to get that function?

Thank you!

Asterisk functions normally have all upper case names and the matching code for them is case sensitive.

Most likely it’s a case sensitivity issue as David mentioned.

However just in case you still have a problem the VOLUME function is located in the func_volume module. By default it should be selected, and enabled in menu select. You’ll want to ensure it’s being “loaded” as well in modules.conf.

The following command issued from the Asterisk CLI will tell you whether it is currently loaded or not.

*CLI> module show like func_volume.so

Hello Kharwell and David,

Thank you VERY MUCH for the info. Although I got 1 more issue with the migration from Asterisk 11.3 to 16. I have 2 wireshark capture, I kind don’t want to post here publicly, do you guys have an email where I can sent to?

So the issue is VERY wired. I have 2 capture, 2 identical calls from our asterisk (same asterisk) to ComExchange (which is an soft pbx used for Hotel based on Asterisk software also). These 2 equipment (ComExchange) belong to 2 of our customers. So these 2 call looks identical, EXECEPT is going to 2 different ComExchange (same software version and sip settings).

These capture are done in the OS of our Asterisk. One call I would see 2-way audio, no issue. The other call, I don’t see Asterisk send back any RTP. Can I email you guys the capture and you can take a quick look? I don’t mind pay for the support.

By the way, Asterisk 16 is in an VM in Azure. I used Ubuntu 18 for the OS.

Thank you!

The other weird thing is that, on the “no audio” call, if I send the call to the old Asterisk 11.3, no issue, 2 way audio.

The call flow is following:

Some one call a phone number à Inbound Carrier (e.g. Verizon) à our OpenSIP à out Asterisk 16 à Customer’s ComExchange

Both good and bad call have the identical call path, it is going through the same OpenSIP and Asterisk 16 butt 2 different customer’s ComExchange.

Looking at the INVITE and 200OK, they all look the same except the IP address and the “TO” header.

It is not an firewall issue because I don’t even see packet going out of Asterisk 16 (since I did the packet capture in the Ubuntu OS).

Thank you!


Never mind, I think I found the issue, although not sure why it is happening. So on the bad call, the ComExchange (asterisk based hotel pbx) is not giving us the 180 ringing and the RTP (media) packet.

Normally, it should be:

We send Invite

They send 100 trying

The send 180 ringing

The send 200OK

We send ACK

And then media start to flow

On the bad call, for some reason it was:

We send Invite

They send 100 trying

They send 200OK

We send ACK

We send the media to them

but they never send us the RTP thus I don’t see RTP going to the carrier.

But the weird thing, again, is that if I send the same call to the same customer via Asterisk 11.3, they do everything correct and have media. I compared the INVITE of both call, no difference. So maybe something in their firewall, I’ll have to work with them…

This topic was automatically closed 30 days after the last reply. New replies are no longer allowed.