PBX BETA Testing (Will be Available for Resellers)

[GreenLantern]

Are virtual/forwarding numbers "routable" within the cloud pbx?

In other words, if a user has a main number A, and a forwarding number B, can the cloud pbx route calls to number B to a different IVR or extension?

Thx

[uf_shane]

Are virtual/forwarding numbers "routable" within the cloud pbx?

In other words, if a user has a main number A, and a forwarding number B, can the cloud pbx route calls to number B to a different IVR or extension?

Thx

My testing with this is yes, but I will let Brandon confirm

[uf_shane]

Are virtual/forwarding numbers "routable" within the cloud pbx?

In other words, if a user has a main number A, and a forwarding number B, can the cloud pbx route calls to number B to a different IVR or extension?

Thx

Phone Numbers

This page allows you specific what extensions phone numbers are routed to if the phone number is not routed to a specific extension when the call comes in or if it comes in via a SIP registration.

Phone numbers are matched based on the phone number in the SIP headers.

If no match is found, the phone number will be routed to the default SIP extension configured under profile.

[uf_shane]

Sent a feature request today for a Device Tab showing all of the registered devices on the pbx.

[GreenLantern]

My testing with this is yes, but I will let Brandon confirm

My testing suggests that forwarding numbers (extra numbers under a single main account) currently are not routable independently from the main number.

I think having separate accounts/trunks/interconnections would work, but now it looks like we can only add a single interconnection.

Not sure if that is a bug, or an intended limitation. But allowing only 1 trunk per pbx is a big disappointment.

[VOIPoBrandon]

Hello - to answer the question on aliases and PBX routing.

Aliases are a high level conversion that happens almost before any other routing.

So when our switch receives a call to an aliased phone number we convert it to the "true" phone number near instantaneously.

This means ultimately that if you were to forward your primary phone number to say sip:uri@pbx.domain, the alias would follow this same exact path.

So there is no independent separation between SIP URI possible in this implementation with aliases.

If you need independent routing the easiest solution is to set up the desired phone number(s) into a cloud account, in which you can route them to any desired SIP uri (i.e. directly to an extension, or to sip:yourPhoneNumber@your.pbx.domain.com and setup a phone number mapping underneath PBX).

Let me know if this makes sense, thanks!

[uf_shane]

Brandon,

Does the PBX support Message Waiting Indicator for voicemail boxes?

[CasualObserver]

Does the PBX support call parking and intercom functionality? Freeswitch has the capabilities and I see earlier in the posts you mentioned that is the platform for this.

Also, any progress on documentation for this? I would think that would be extremely high in your to-do list since you are charging for this in open beta. Star codes, features available etc would be nice. I see several such as mail access and recordings have been posted in this thread already.

Thanks!

[VOIPoTim]

Does the PBX support call parking and intercom functionality? Freeswitch has the capabilities and I see earlier in the posts you mentioned that is the platform for this.

Also, any progress on documentation for this? I would think that would be extremely high in your to-do list since you are charging for this in open beta. Star codes, features available etc would be nice. I see several such as mail access and recordings have been posted in this thread already.

Thanks!

We have been gradually adding documentation to the control panel pages consistently and will continue to gradually add more and more. It is a big priority.

In terms of intercom and parking, we can add those to our suggestion list for future consideration.

[chevyman]

Tim

I have found the Voice Mail system to be confusing. Say your sip extensions are numbered in the 300's, so you have the VM extensions in the 800's so ext 301 is using VM 801, but if someone calls ext 301 and get that ext VM, the generic message says "person at extension 801 is not available, record your message at the tone..." this is confusing for the caller, because they were calling extension 301.

There must be some way to link sip extensions with VM extensions. so the generic message would say the sip extension number and not the VM number.

thanks