Incoming calls SIP

[gunners]

We are changing over to SIP trunk from our current PSTN lines.
Currently if one of the PSTN lines are engaged then it automatically gets forwarded to net next line.
How do you do this with SIP trunks?
Is it done with the provider or on 3cx server?

 

[ucs1]

Hi Gunners,

**Disclaimer: I've been told I should never try and explain the difference to anyone

You currently have PSTN lines setup in what I'd assume was a primary number and steppers (additional lines with their own numbers) each one requires a physical copper connection back to the exchange and each POTS line can carry 1 call.

Lets assume your have 4 lines and 4 numbers.

Line 1: 1234561
Line 2: 1234562
Line 3: 1234563
Line 4: 1234564

Think of SIP trunk as a raw pipe with dissociative identity disorder (or what I like to call DID). Lets also assume that your SIP Trunk has 4 Channels (the ability to carry 4 simultaneous calls at once)

Channel 1: At the point that someone dials 1234561 - Your first channel leaps into life and says "YES I'll take that call" - Lets assume this call continues for the whole example. This call will follow the rules / routes you have configured within 3CX
Channel 2: When another person then dials 1234561 - Your second channel bounds up and down saying "PICK ME, PICK ME" and at the same time accepts a call for 1234561 - just like the previous call this will follow the same rules / routes you have configured within 3CX
Channel 3: The third call comes in on 1234564 - Your 3rd channel takes on the role of your old Line 4 and the call is then served again as desired.
Channel 4: You want to make an outbound call to 2345671 whilst all of this is happening. Channel 4 Takes on the role of your outbound number (or Line 4 in the PSTN example above)

Although you have a Trunk of 4 Simultaneous calls they all have the capability of acting as one of those 4 Lines you used to have.

As for the routing of calls within 3CX you can create IVRs, Ring Groups and Queues to direct the traffic as required (or a direct connection to specific extension if you so desire)

A trunk is more like ISDN where we separate "numbers" and "channels". A channel can have one call (either outbound or inbound) in it at a time and the "numbers" are only there so you can work out who they want to call. So with sip a call will just arrive as the next "channel" available in the trunk (up to the limit your SIP provider has set) and you just look at the number being called and then send it to wherever you want to.

This gives you massive flexibility and direct control of when and how you send calls to different places instead of paying your old telco tons of money for the chance to do half of it.

 

[lneblett]

SIP trunk providers can usually be separated into two camps:
1, Single call path
2. Unlimited call path

A single call path provider is more akin to PSTN in that one call is allowed at a time using one number. An unlimited call path provider is where one number can handle as many calls (using the same number) as your bandwidth or pbx license will allow (whichever is less). With unlimited, you can add DIDs to enhance the routing or provide "private" lines for certain areas/people.

A single call path provider will charge a monthly fee per line. An unlimited call path provider will charge on a per minute rate or possible package minutes into bundles.

Usually, rollover on an incoming call, from a single path provider, is handled by the provider....assuming that the callers are dialing in using the same number.

An unlimited call path provider simply sends the call down and the PBX will handle it. A poor analogy is to think of it in the framework that you have one internet pipe and one service and one IP. In the office you have a number of people all sharing that one resource; yet all are working or looking at different items at the same time. SIP has functionality that allows a similar flow in that each call is uniquely identified so that multiple calls can be handled by one PBX and be directed to multiple extensions so that each extension can have its own call in progress. Much like John can be on AOL, Mary on Google, etc.

So, the answer to the question depends on the provider and the call path supported. Hopefully, you have or will select a provider from the 3CX supported list as this will lessen the likelihood of issues later on and the provider should be able to assist in giving you their method of busy call forward rollover.

As an aside, you should also determine how you want your outbound CallerID to look as this may also be influenced by the method and/or provider.

 

[gunners]

Excellent, Thank you that actually makes sense now

 

[brentkhack]

I had the same exact question but this is the answer I needed. So to make sure, I have nexVortex as my SIP provider and get 5k inbound and outbound mins per month. So by the example provided I would assume it is unlimited. So I should be able to port my one marketed number over to nexVortex and then I can make up to 8 calls (per license) at the same time with that one number right?