Understanding trunks and inbound routes

i have a trunk with 2 channels (nb1 and nb2) and 1 inbound route.
if i set the DID of my only inbound route to nb1, the trunk will use the 2 channels for the incoming calls, right ? i mean i can have more concurrent calls than if i have only the channel nb1 in my trunk ?
Thanks for your help !

This sounds like FreePBX terminology. Asterisk doesn’t use the terms trunk or route. DID seems to be a term used by SIP sevice providers, so I assume you are using SIP. The SIP RFC does not contain the word “trunk”.

For SIP, Asterisk channels are created dynamically and there are no hard limits. However you do not be using “channel” in a sense understood by Asterisk. The provider might use channel in a different sense and might impose limits.

If you are using FreePBX, you might get more luck on https://community.freepbx.org/ but you should still clarify what “channel” means to you, as I don’t think FreePBX uses the term in its user interface, so it is only meaningful for power users, customising it with knowledge of Asterisk.

Thanks for your answer !
i thought freepbx was just a gui for asterisk…
ok i will try the freepbx forums so people can understand the words i use.
thanks for having taken time to answer me :blush:

FreePBX is a (single tenant, but otherwise feature full) PABX implemented, in part, using Asterisk. The GUI is for the PABX and doesn’t directly control Asterisk. It creates various abstractions, including incoming routes, and DIDs, which are not explicit in Asterisk, and uses some terms differently from Asterisk. DIDs are just extensions, in the Asterisk sense, as far as Asterisk is concerned.

Thanks !!!

have a nice evening

maybe i can try to re-ask my question with other words.

is it possible to have 10 concurrent calls on 1 single sip number ?
what is the topic name in asterisk to do so ?

Thanks a lot :slight_smile:

Asterisk has no limit on how many calls can be handled concurrently on a
single number - it really doesn’t care anout the number, other than that’s how
Asterisk knows how you want to process the calls.

You configure a context with the inbound SIP connection/s pointing to it, and
you define the extension corresponding tothe incoming number. It doesn’t
natter whether multiple calls come in tothe same number from different SIP
connections or the same one - provided whatever is making those connection in
to your Asterisk instance supports placing multiple simultaneous calls.

After that, it’s up to your dialplan what happens to each call and how it can
be handled concurrently…


Asterisk imposes no limits on the number of sessions with the same request URI. Any limit on that basis would have to be explicitly coded in the dialplan, e.g. using GROUPCOUNT().

I think there may still be a mechanism to limit the number of sessions associated with a particular endpoint (which is probably the closest equivalent to a FreePBX trunk, except that endpoints can also be FreePBX extensions. I believe those mechanisms are deprecated.

It is possible that FreePBX provides options to apply request URI and endpoint based limits, but the former would be the result of its generating dialplan that achieves that result.

The service provider may limit the number of sessions, based on various properties of the call.

There will be global limits based on the availability of system resources, but these won’t normally distinguish between endpoints (including between those that FreePBX calls trunks and those that FreePBX calls extensions), or between particular request URIs.

Also that many provider pass what they call the DID in the To header. The contents of the URI part of the To header has no intrinsic meaning to Asterisk. In such cases, FreePBX will use explicit dialplan code to extract the value and use it as though it had been the request URI. Other headers are sometimes used, but they have to be handled with custom code, even in FreePBX.

FreePBX doesn’t care from which “trunk” type endpoint a DID was obtained, although the way it determines the value to use may vary between endpoints.

Thanks David and thanks Pooh for your answers.
i clearly see my actual knowledge around sip is largely not enough :smiley:

my problem is that i read on many pages that to handle n employees in an office you have to have n/3 channels in your trunk.
i understood a channel as an sip line or number.
my provider proposes single line sip or trunk (single lines in an entity).
i thought then that for my 30 employees i would need 10 sip lines gathered in one trunk.

but as you said this should be asked in a freepbx forum.

i learned a lot with your answers !
Thanks !!

  1. Will your SIP provider allow 10 simultaneous (inbound and/or outbound) calls?
  2. Will all inbound calls be identified as the same DID?
  3. Read about how to configure FreePBX with the above answers.
  4. Ask on the FreePBX forum.

Thanks !
i will do that

These are basically commercial constraints applied by the provider.

This may help you understand things a bit better:

Asterisk uses the term “context” to mean the part of the dialplan which starts
processing a call which comes in on a SIP connection. Several connections
(eg: from different providers, or from different telephones) can point to the
same context.

An “extension” is the number (or pattern, also to match a variety of numbers)
within a context which determines where Asterisk starts processing the
dialplan commands.

A “channel” is an internal Asterisk thing, which most people don’t need to
think about, but essentially means one communication to one other device or
system (eg: a telephone or a SIP provider). Nearly all “phone calls” (human
terminology) in Asterisk consist of two channels, connected to each other
(“bridged”) by the dialplan.

External providers of SIP connections often use the word “trunk” to mean a SIP
connection which can carry multiple simultaneous calls, and the maximum number
is often determined by how much you pay them. Asterisk doesn’t use this word.

SIP providers often use the term “channel” to mean “one connection going over
a trunk”, so if a trunk has 10 channels, it can carry 10 calls at the same
time (either incoming or outgoing, as far as your Asterisk system is

FreePBX also has its own vocabulary, which is mostly a bit of a mixture
between the above, but specifically it uses the term “extension” to mean “a SIP
endpoint such as a telephone”, which is probably the most different usage of
the same word from the way Asterisk means it.


Thanks a lot Antony !
it’s more clear and i’m happy not to have understood everything wrongly :slight_smile:
i’ve just asked my provider for the number of simultaneous calls in its trunks. let’s see how much it wants to charge.

Thanks to all of you for having helped me and even if i should have asked freepbx forum.

Long live Asterisk and its community !

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