Is the asterisk support the class 5 services?

I need to know if the asterisk supports the class 5 services

Doesn’t that refer to outdated technology? In a way Asterisk can act as a class 5 or 4 service, but generally proxy servers like Kamailio are more like class 4 services.

Thank you for you prompt reply :slight_smile:

No, it refers to a level of service that is provided. A Class 4 Switch is a PSTN switch. It’s only job is to handling the ‘switching’ of calls between the network and the PSTN. It basically provides dial tone.

Class 5 services refer to what is considered Subscriber Services. This would be the services that a PBX or other media/voice applications provide. Such as voicemail, call waiting, vertical (*) codes and others.

Asterisk is not, in any way, meant to be a Class 4 system as it is a B2BUA. It can provide Class 5 services at most.


Thank you EkFudrek and BlazeStudios, for your replies.
I need to know then, how a SIP client , in its INVITE message, can enables the class5 services in Asterisk ?

It doesn’t. If you want them to have voicemail, you let them have voicemail. You want them to have specific feature codes/star codes, you create them. Nothing in the invite does this. It’s how the system is setup and that depends on you.

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This will help for a better understanding

I think the problem here may be a failure to understand that Asterisk is a toolkit for creating PABXes, not a PABX.

Actually looking at the definitions, I would say that Asterisk can construct class 4 switches as well as class 5 switches (and hybrids).

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Well let’s been frank on something. Asterisk is not meant to be a switch, it’s a B2BUA. No real softswitch is going to have what akin’s to a two-legged call for every call. Plus all the other switching features Asterisk lacks.

So as I said before, at best Asterisk can be used for Class 5 switch features because that is where the IVR, voicemail, ring groups/hunting, etc all would live.

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Asterisk was meant to be an analogue and ISDN switch, and that is how its internal deep structure works. SIP was bolted on. The B2BUA concept is SIP terminology, where the other option is proxy. Switch covers both.

And those days have passed. So you’re saying that you would choose Asterisk as your switching solution versus all the other Telephony switching appliances/software that is available? Because serious providers wouldn’t.

I would certainly consider it for a tandem switch in a medium to large enterprise, although, typically, you would not have dedicated tandem switches.

Also, I suspect a lot of people still use Asterisk with ISDN and analogue, as Digium doesn’t seem to have closed down that part of their product line.

ISDN, PRI, T1, et al aren’t going away anytime soon, I understand that. However, that is more due to the existing infrastructure. At my time with the LEC we went from using DS1/DS3 circuits for A to Z to Fiber. We went from switching over TDM/SS7, etc to IP for interconnecting with other LECs and carriers. Not only did it bring down the peering/usage costs, it brought down the infrastructure costs while gaining improvements and features. Even over in the EU right now they are making mandates to push carriers to fiber, etc over copper. BRI’s, POTS, ISDN style connections are being upgraded to new tech.

There is still a market for analogue and digital equipment but it’s dying quickly. You used to be able to swing a dead cat and find a vendor that had FXS/FXO one port gateways for SIP/VoIP and now there’s like Grandstream. While others are still doing those gateways, they are doing them in multi-port appliances because the single port need is really gone.

Even now most of the installs I do for clients involve either completely dumping the analogue/digital stuff or moving to using gateways so the actual PBX can be VM’d and not have to worry about cards, drivers and physical hardware issues. With hotels, if their legacy PBX loses a card it is actually cheaper for them to go IP PBX with gateways vs new hardware because even companies like Mitel don’t want this crap out there anymore.

In fact there is some sip server requires a suffix added to the invite number to enable needed services.

I’m not sure what that has to do with any of this.

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