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
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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 confirmQuote:
Originally Posted by GreenLantern
Phone NumbersQuote:
Originally Posted by GreenLantern
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.
Sent a feature request today for a Device Tab showing all of the registered devices on the pbx.
My testing suggests that forwarding numbers (extra numbers under a single main account) currently are not routable independently from the main number.Quote:
Originally Posted by uf_shane
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.
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!
Brandon,
Does the PBX support Message Waiting Indicator for voicemail boxes?
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.Quote:
Originally Posted by CasualObserver
In terms of intercom and parking, we can add those to our suggestion list for future consideration.
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
Quote:
Originally Posted by chevyman
i see they added 'Default Greeting:' but no matter what setting i try it still says ext 801 and not 301.
EDIT:
i just was trying out the IVR extension and it times out to VM and the 'Default Greeting:' is working on it but not on the SIP extensions. but now my IVR greeting isn't playing since the last update. I guess its time for a trouble ticket.
Saw that IVR not playing also, try setting the fields under it to non-zero values. Maximum timeouts and / or Maximum failures fields is what did it. After I did that, it started working.
Also, would be nice to be able to use shared line functionality. Freeswitch has it, but it needs to be enabled to work. For small businesses that can be handy along with the call parking etc I already asked about.
Thanks!
I was wondering if in the future sometime, resellers will have the option to control the number of extensions a client would be able to use, so that it would be easier to charge per extension, rather than have to monitor the clients account to see how many they are setting up.
Here's a simple tweak request...
Could we get some more IVR timeout options?
10 seconds is currently the minimum timeout.
Would like to see 1, 2, 3, 4, 5, 6, 7, 8, 9, 10, 15, 20, 30.
Can't really imagine a scenario where you'd want more than a 10 second timeout.
You generally don't want the caller just sitting around very long.
This has been completed.Quote:
Originally Posted by GreenLantern
Is there anyway to show in the SIP Extensions if a device is registered to that Extension. Like in the reseller account under 'Connections' where you get the MAC, IP and Device information.
I noticed a new extension type today "Bind Meta Application"
Anyone else having issues with unregistering a number in the pbx? Even removing the number is not working for me. Still shows as registered in the reseller account.
Pop a ticket in, Brandon has been pretty quick with me as we were investigating a different issue with registration
This feature has not yet been fully pushed out just yet - however is coming very soon, stay tuned....Quote:
Originally Posted by uf_shane
Hint: allows in-call actions to be bound to a given call flow, i.e. call transfer, call park, call pickup, etc.
Sweet can't waitQuote:
Originally Posted by VOIPoBrandon
Hi guys. Would it be possible to put this bit of info somewhere on the voicemail page? would really help. thxQuote:
Originally Posted by VOIPoTim
I just tried to add another gateway and got this error "You are only able to create a single gateway at this time".
Why is this?
I thought the only restrictions was 10 connections?
A gateway is the SIP provider where your outbound calls are sent to. You can only have the outgoing calls go to one provider at a time (us or another SIP provider).Quote:
Originally Posted by chevyman
Then how do you do inbound numbers, so some sip extensions can also have direct dial outside numbers?Quote:
Originally Posted by VOIPoTim
You have a few options for adding inbound numbers:Quote:
Originally Posted by chevyman
1) Order the numbers in the PAYG section for $1/mo + $0.01/Min and select what extension you want them to route to.
2) Point the number to the SIP address of the extension you want it to route to directly.
Go here for more detailed info:
Interconnection - Incoming
My PBX has no "PAYG" or "Interconnection - Incoming". all i have is Interconnection - GatewaysQuote:
Originally Posted by VOIPoTim
Ah this is a reseller PBX then right? That's for our direct customers. In your case, you'd want to just route the DID to the SIP address of the extension you want it to go to. So you'd set up the DID in a resold account and then use the SIP forwarding function.Quote:
Originally Posted by chevyman
Is anyone else having trouble with the Cloud PBX?
Starting Wed Feb 5, and through today, Thur, Feb 6, we're getting tons of dropped calls on all our reseller PBX's.
This even happens on internal extension to extension calls, which do not go out on the trunk.
I can duplicate this by creating an echo extension, then dialing that extension. The audio will drop as quickly as 30 seconds, or I may get 7 or 8 minutes. But the audio drops every time. This has been duplicated from several different cities/states, for at least 5 different cloud pbx customers in the last 24 hours.
Just trying to see if it is something specific to our customers, or if it is a problem with the reseller cloud pbx.
Thanks
i tried it with echo and it dropped at 1min 25sec and second call at 30sec.Quote:
Originally Posted by GreenLantern
i'm on reseller pbx.
i'm not sold on this pbx for any type of production use.
We are working to address an address and isolate it where we have a user either maliciously or just with very misconfigured devices sending large amounts of invalid packets to the PBX system that are causing it to "segfault". Our monitoring is then detecting the failure and automatically rebooting but then the attacks whether intentional or non-intentional keep happening. We're working to isolate them, block them, and then put in workarounds to handle that invalid traffic in the future to prevent this.Quote:
Originally Posted by GreenLantern
where is the "SIP forwarding function" at?Quote:
Originally Posted by VOIPoTim
Anywhere you can put in a phone number as the destination, you can put a SIP address. Here's the format: sip:123@host.server.comQuote:
Originally Posted by chevyman
So to forward to extension 8482 on demo.voip949.com, you'd do sip:8482@demo.voip949.com
sip:extension@PBXdomain
You can direct it right to the specific extension you want this way whether it's a user extension, a call queue, a group, etc.
I didn't know the forward number could be a SIP address in the reseller account, thanks.Quote:
Originally Posted by VOIPoTim
If i have a 6 line phone (Grandstream gxp2200), having line#1 as main in-out line and lines 2-5 incoming only, all going to IVR. Is there a way to know when a call comes in, which of the 5 lines it came in on.
I don't have that phone, but normally, all 6 lines would be registered with a single extension registration. Incoming calls on line 1 would flash the line 1 indicator light and ring your selected ring tone. Additional calls would flash the next indicator light and use the same ringer tone.Quote:
Originally Posted by chevyman
If you want something like different ring tones, one way is to create 2 or more different extension accounts for a single phone. Register the phone to several different extensions (1011, 1012, 1013... 1016). Then assign line indicator buttons/lights 2-6 to different extensions instead of all to extension 1 (account 1). Now you can set different ring tones to each line/extension. This lets you get specific about routing different ring groups or queues to specific extensions, so you get different ring tones based on what menu option your caller chose. You might want to limit yourself to just 3 different extensions, so each can have 2 lines on your phone.
The above is a lot to keep up with, but it might do what you need, given the tools available.
More established pbx systems have more options available such as "caller id prepend" and "agent announcement" that could help you out. I'd love to see these features implemented in the Cloud PBX at some point, especially "agent announcement", which lets you assign different pre-recorded messages to your Ring Groups, then plays the corresponding message to an agent when they answer a Ring Group call.
Anybody else having issues with their phone number not forwarding to their pbx? Mine is set up to forward to ext.400 in my pbx. Ever since last night, all I get is ringing on the number that's supposed to forward to the ext. 400. I checked everything in the pbx and all is fine.
Yes, it seems to have started around 4pm Wednesday afternoon. I opened a ticket and was notified Thursday night that it should be fixed now.Quote:
Originally Posted by wingsohot
But we're still experiencing issues with outbound calling. Only about 1 in 4 seems to work properly.
Brandon just replied to your ticket.Quote:
Originally Posted by GreenLantern
So far we've been able to find a number of bugs/scenarios during the PBX BETA that could only be discovered in real-world usage and we appreciate all BETA testers helping us test and find those bugs now and being patient as we work through them.
We think all current PBX issues are now resolved. If you continue to have issues, please update your ticket.