Use last channel to forward to VM on a sip trunk?

First post here. I’ve searched high and low and cant quite figure out where to start.

I have a client who has FreePBX and a provider with 4 sip trunks.

They would like use of the 3 trunks with the 4th trunk reserved for transferring the 4th caller to VM. They want to avoid busy signals when possible. Is there a solution for this?

If I could post a picture of my German Shepherd looking at me, I would.

Unless your customer has a really, truly bizarre setup, SIP trunks are usually unlimited in the number of calls they can carry. The only reason people have more than one is to support redundancy or to support a specific kind of call to a special recipient (I have two, one for normal traffic and one for “forwarded” calls to my cell phones).

Second, Voice Mail is handled in the server, so it would never tie up “another” SIP trunk. It routes the call to VM and you’re done with it.

So, Lucy, you got a lot of 'splaining to do… :slight_smile:

Tell us what you understand the system’s setup looks like and we’ll try to help you understand. I think I might have an idea of what you really mean, but what you said is not what I think you are going to end up telling us you meant, and even then, VM won’t take another “trunk”.

For sure. The provider calls them “Trunks”.
To me they are “Channels”. Is that what you mean?

I’d like to reserve the last “channel” in the (one) “trunk” in asterisk for forwarding that 4th inbound call to voicemail. The provider only allows one call path for each (there term) “trunk” .

Channels and Trunks are not terms used by VOIP providers. Actually, no, that’s wrong. Let me rephrase that - a trunk is the logical path by which your VOIP channels are delivered to your PBX, but there are almost no providers that use those in a 1-for-1 configuration. Most providers deliver any number of channels over your “trunk” (which is logical) and is basically limited by the network bandwidth available.

When used by PRI or POTS vendors, they are always used in terms of "lots of channels (up to 23, 24, or 30 depending on location) are sent over a single trunk. No one would use a trunk with only one channel on it. They would call it a “line”.

There is never a time when you would switch a call from the “trunk” or “channel” it’s on and put it on a different one to send it to Voice Mail. The answer to your initial question is always going to be “No, that doesn’t make any sense” - at least, until you can get a clear understanding of what it is you are working with.

What is your current PBX? Manufacturer? Model? Interface? What kind of cable? Are you using FreePBX? Post your Trunk Configuration (passwords hidden, of course).

I understand that you might not understand what’s going on right now, or that the terminology is a little opaque. We’re (almost) all good with that - tell us what you have and how you get it, and we should be able to piece together what you’re really working with.

Now, assuming you are working with FreePBX - when a call comes in, if it goes to VM, it does so as a consequence of a series of control steps on the PBX. I never leaves the control of the server, so it never uses another trunk line. There’s no way to transfer the call to another line without tying up both lines, which defeats your “no busy signals” requirements.

I’d like to help you, but the universe of VOIP is vast and the terms you are using mean very specific things and don’t sound right in the context you are using them.

You are correct. FreePBX.
They have a SIP Provider who provides them with… service. I’m merely quoting what this provider has chosen to call the service. They have purchased what the provider calls “SIP Trunks”. They have purchased 4 of these “SIP Trunks”. I have what Asterisk/FreePBX labels as a Trunk configured in my PBX with all the details of this provider. There are no issues with this setup currently.

My only question (regardless of how many lines/trunks/channels/call paths it takes up is; Is there a way to put a limit on the amount of calls that come in and/or reserve 1/2/15 lines/trunks/channels/call paths for voicemail. The goal is to avoid a busy signal. I am a server admin. not a PBX/voip/asterisk guru by any stretch.

Does this help clarify at all? I feel like I’m being redundant or pedantic. Not intended. Thanks again for any help.

SIP trunks are only limited by the provider. If you have a one channel per trunk setup, there is no way to fix this. How many trunks do you have set up with your provider? You said four originally, but now you are making it sound like just one. It makes a HUGE difference.

Let me say this part again. SIP trunks are virtual. You’re a server admin, so let’s try this - they are UDP ports open on the servers at each end. They are connectionless and are unlimited by the underlying technology. They are only limited (in the technology) by the bandwidth you have available. Asterisk (and the management framework we call FreePBX) is capable of handling hundreds of simultaneous calls on a single “trunk”.

So, if someone is using all four of your trunks and another call comes in and you get a busy signal, you should change providers. There’s no excuse for this kind of customer abuse in the current state of technology. Unless you are mandated by a government or legal body to use a specific ITSP, your setup and your situation are both more complicated and less useful than is acceptable in this day and time.

You keep asking this question about Voicemail. It doesn’t make any sense. There are two use cases for voicemail:

  1. Someone calls and wants to leave a voicemail. If they called and didn’t get a busy signal, they can leave a voicemail. This does not use any system resources that aren’t already being used.

  2. Someone wants to call in and check their voicemail. They shouldn’t be calling in through the main number, they should be using a VOIP phone that can connect to the server and be “inside” the system. They wouldn’t ever touch one of your scarce inbound line resources. You can set up a VPN from the phone to the server and they can take their phone with them wherever they go.

So, neither of these uses any more resources than are already being used for the call. If I’ve missed some other use case, please help me understand what you think the problem is.

A custom context or two that implements the count functions

They (the provider) have sold us 4 SIP Trunks. I was speaking as to how it is displayed in FreePBX as 1 trunk. I am assuming the typical setup from the provider is to setup one trunk inside of asterisk/FreePBX for all 4 call paths.

I had worked in the past with one provider that used busy signal detection and allowed for that call to be forwarded to another destination, but this provider only does fail over routing if the PBX is unreachable.

I was just trying to see if there was a possibility of using the last channel/trunk from the provider to send the caller to voicemail (to leave a message) instead of a ring to the ring group. Another way of explaining what I am trying to do is to cannibalize the last call path and make a rule/route to voicemail instead of just ringing the phone(s).

Thanks for those! I will be reading to see they can be applied to our situation.