Feature requests

[Vali_3CX]

Hi all

If you tested the 2.0 Beta version of the gateway and you have suggestions/ideas to improve it to make it more confortable for you, please post them here - any will be welcome.
However, there is no guarantee it will be implemented.

Thanks for your feedback
vali

[sipero123]

Hi,

I think the option for multiple STUN servers would be great

Another alternative would be ability to set the external IP then stun would not strictly be required.

As you are perhaps aware of other discussions I would like to be able to run the gateway and Skype on a machine due get around the issue of the 3CX PBX being on a HyperV machine that does not support audio and that may not be on the same subnet as the machine running the 3CX skype gateway.

Also I would like to see frequent smaller upgrades rather than having to wait for a longer time with more upgrades. Perhaps 4 to 6 weeks rather than around 3 months would be good.



Jonathan Hamon

[zensoftware]

Hi Vali,

I think expanding the outbound rules in the PBX to make better use of the Skype gateway would be a great feature. If you could specify calls which have alpha characters rather than numeric ones should go out over the Skype gateway.

It would also be useful to be able to forward calls to a Skype client in the same was as you can to a mobile phone or external number. If a call can be sent to me on say my iPhone SIP client that would save the company the cost of the PBX to mobile leg of the call.

Regards,

Emmet.

[Vali_3CX]

thanks Jonathan, Emmet

Emmet, if I remember, some time ago you said something about an issue in gateway 1.0 regarding some conference call - did you tried it also with the new one? May you provide a configuration description to test it myself?

Thanks
vali

[zensoftware]

Hi Vali,

I'll try it later today and let you know how I get on.

Emmet.

[zensoftware]

Hi Vali,

Ok, it works perfectly now. When I call the specified number, it transfers me to the conference server.

Just one drawback though, Skype has no ability to send DTMF tones so I can't enter a conference ID and then press #

Emmet.

[zensoftware]

Me again.

A bit of research has shown me how you can send DTMF tones.

Free call conferences anyone?

Emmet.

[Vali_3CX]

Emmet, Jonathan - I am interested to know your opinion, as experienced users you are, about a possible feature (it might be useful, or not?)

So, test environment describing the issue:
- the gateway has assigned a simple outbound rule - a prefix "sk" and strip 2 digits.
- inbound calls coming from this gateway are routed to my deskphone, 101.
- while not being on my desk, Jonathan called me from his Skype so, when I'm coming back, I see on display 1 missed call from sipero123. As an lazy user I am, I'm pressing dial and, obviously, doesn't work, I get "not found" because "sipero123" is not known, so I have to type "full name" as "sksipero123".

The question is: do you think it might be useful to allow gateway to prepend a user-defined prefix (for each of its ports) to the incoming "from" name? In our case, supposing this prefix is set to "sk", the standard call from "sipero123<sip:sipero123@xxx.xxx.xxx>" will become "sipero123<sip:sksipero123@xxx.xxx.xxx>", therefore when I will want to dial back from my 101 I will have nothing to do more than to press dial button.

Regards
vali

[sipero123]

Hi,

Sounds like an excellent idea to me, but perhaps you need to make it an option rather than just doing it for every port.

You would probably only want it for skype or gizmo ids and not for others. This way it would already know the correct route to use for the return call.

Its importance I think is increased by the fact that for many phones it is not a quick process to dial names rather than numbers other than by using speed dial.



Jonathan Hamon

[Vali_3CX]

but perhaps you need to make it an option rather than just doing it for every port.

Hi, Jonathan
I'm thinking to every port because each has a corresponding PSTN gateway on the PBX - therefore a corresponding outbound rule.
So:
port 10000--- PSTN gateway 10000---outbound rule "sk" -> prefix to set will be "sk"
port 10001--- PSTN gateway 10001---outbound rule "ak" -> prefix to set will be "ak"
and so on.

About phone numbers, suppose you have an inbound phone call to gateway's Skype from, let's say, 00xxxxxxx. With prefix added, your 101 will receive a call from sk00xxxxxxx, so it will be able to call back to the same "route", the Skype.

Regards
vali

[zensoftware]

Hi Vali,

This sounds like a very good idea.

But, how would it work for calls that come in via the Master Skype account?

Emmet.

[Vali_3CX]

But, how would it work for calls that come in via the Master Skype account?

Hi, Emmet

Good question
The answer is, in my opinion, Master Skype account will behaves as user want, and it has three choices:
- 1. inbound call cannot be dialled back
- 2. inbound call can be dialled back through master if master is not busy
- 3. inbound call can be dialled back through a slave.

Suppose we have a configuration with one master and two slaves, having PBX-outbound rules "ma", "s1", "s2" respectively, and a "caller" Skype caller.
Then, from gateway's prefix point of view
- 1. inbound call cannot be dialled back
master prefix set to blank. Phone will receive call from <sip:caller>, therefore it will not be able to dial it back since an outbound rule is missing, so it has to diall the full name (it's actual behavior of the gateway, in fact, regardless master/slave type);

- 2. inbound call can be dialled back through master if master is not busy
master prefix set to "ma". Phone will receive call from <sip:macaller>, therefore it will be able to dial back through master only this one is not busy - according to master behavior, when it is alone, or all its slaves has in-progress calls and, obviously, when itself is not busy.

- 3. inbound call can be dialled back through a slave.
master prefix set to "s1" or "s2", say "s1". Phone will receive call from <sip:s1caller>, then it will be able to dial back "diverted" through one of slaves, if this one is not busy. However, if "caller" Skype account knows only about "master" Skype account, it might wonder who's "slave1" skype account calling him. Also, in case of phone calls, "master" Skype account most probably has a different credit than "slave1" Skype account. But, on the second thought, this two issues would appear even master channel would be an "outbound master", because in this case it - not the prefix - would be the one dispatching <sip:macaller> to one of its slaves.

I cannot imagine other combination right now

Regards
vali

[arpa]

As a feature request for Skype gateway , i would like to check which skype accounts would be slaves to a specific master account and wich will be alone. For example i have i skype account an i dont want to be group with the "business" skype accounts. Or to check 3 slave skype accounts to belong to 1 master and 2 others slave to another master account etc for sales, support ...

[Vali_3CX]

As a feature request for Skype gateway , i would like to check which skype accounts would be slaves to a specific master account and wich will be alone. For example i have i skype account an i dont want to be group with the "business" skype accounts. Or to check 3 slave skype accounts to belong to 1 master and 2 others slave to another master account etc for sales, support ...

Hi
Arpa, gateway, even in its actual Beta form, is (very) flexible. I suggest you to try following configuration:
- configure 6 Skype ports on your gateway, with no master, say 10000-10005.
- configure first three, 10000-10002, to use the same Skype account, say "ArpaSales", and the next three, 10003-10005, another Skype account, say "ArpaTechSupport".
- configure, from File/Settings/Advanced, "Ring group behavior on inbound Skype calls" (check that option)
Now, supposing on your PBX you have phone extensions 100-102 belonging to "sales personals" and extensions 103-105 belonging to "technical support personals", on the PBX:
- on corresponding PSTN gateways 10000-10002 assign them to extensions 100-102 respectively, and 10003-10005 to extensions 103-105.
- on corresponding PSTN gateways 10000-10002 set outbound rules to allow only outbound calls from 100-102, and for 10003-10005 - from 103-105.
Then, start the gateway.

Now
First inbound Skype call to ArpaSales will "ring" 10000-10002 gateway's ports and, obviously, 100-102 "sales personals" phones. 101 will pickup the calll, 100 and 102 will stop ringing and will become available, as their gateway's ports. Another Skype call to ArpaSales will ring. Since 10001 is busy, will ring only 10000 and 10002 and their phones. 102 will pickup the call. At this moment, two sales personals are talking simultaneously, and 103 is available. Now an inbound Skype call to ArpaTechSupport is coming, so all 10003-10005 gateway's ports will ring so their "technical support phones" 103-105. 103 technical support guy will pick up the call, so 104 and 105 - and 10004 and 10005 - will stop ringing becoming available. And so on.

Hope this idea will help you to solve you problem. On the version I have (I mean, next release) the inbound Caller ID is fixed, so all phones will receive Skype caller's first/last name, if available, or, if not, skype account as CallerID, so you may combine better gateway's capabilities with PBX's ones to meet your demands.

Please tell us how is going, if you test this.

Regards
vali

[Elysium]

I think it would be a good idea to have something like this.

I'm from Europe and let's say I go to Thailand to a vacation. When on vacation people are calling me from time to time which means my mobile phone bill would be very high.

My solution is as follows. I have Skype installed on my phone and I can call my 3CX installed at home anytime using WiFi. I can route this call to any extension or a digital receptionist. Now the problem is that I'd like to call one external number (out of the PBX) at the moment I'm connected to my PBX. I can only forward my call to a predefined phone number for the moment but I would like to type a number that I would like to call.

Yes I know I can use Skype to make phone calls, but this example was just for easier understanding of what I'd like to do.