So I see a lot of discussion about how to use FreePBX with softphones and some talk about specific phones. Unfortunately, the thing I’m trying to do is use the already-existent Android smartphones of the various people in the company as their phones in the PBX system, for handling incoming calls to the company’s number and company voicemail.
I do have a trunk/number through SIPStation and my server is running 2.6.32 off of the default FreePBX Distro.
Any help appreciated! 
if i read your request correctly, what you are in essence trying to do is register your smartphone directly with the SIP server. This would actually be running a SIP softphone client on the phone. if that’s it, you simply treat it as a softphone, which it is.
If that’s the only way to do it, that’d be… unfortunate, I guess? Especially in terms of battery life.
If I understand correctly what you’re saying, there’s no way to register an external, non-softphone number (e.g., to just have a phone whose number owns an extension and which can call in to get voicemail etc in the same way a softphone can)?
If that’s the case, is there at least a way to set it up such that when an extension gets dialed, it dials an outside number (so if someone presses 0, it goes to my cellphone or Google Voice number)?
Have you tried a quick visit to:-
yet?
I have indeed! Well, actually, no, I tried a long series of visits to the documentation page over the past weeks and, more intensely, the past few days. If there’s something that addresses this question, then I missed it; I did notice that how to use a SIP phone is covered, as well as specific recommendations for specific SIP phones to use, but not how to use a hardphone for the same purpose.
Any hardphone that is designed to use one of the VOIP protocols supported by Asterisk will work.
Your’s does not and being a cell-phone (mobile) is not considered a hardphone by this community.
Phone numbers are not directly associated with phones, they need to be routed to hardware that responds to their signaling. You can often arrange to have calls rerouted and your phone can support “softphone clients”, normally iax2 or SIP.
Maybe you need some more general background reading:-http://www.voip-info.org/
start with the appropriately named “Getting Started” section . .
Entertainingly, even after finding the Getting Started page among the eyebleeding explosion of advertisements and colors, the information isn’t there. But you’ve answered my question (to wit, that it’s not possible) and I thank you for that, dicko.
Now, if only the guidelines for “How to do external calling through Misc Destinations” that I have found on this site actually worked, I’d be set!
Why the angst?
http://www.freepbx.org/support/documentation/module-documentation/miscellaneous-destinations
put in a mnemonic and your phone number and it is available as a destination all over FreePBX, IWFM
maybe I misunderstood what you want to do. Do you simply want to route calls to a cellphone over the PSTN?
Yes. I want to route calls to cellphones. I don’t care if it’s over the PSTN or over IP; I need the system set up such that a call to the service results in an account manager’s cellphone ringing.
@dicko: The misc destination page says that it allows you to dial “anything you could dial from a standard extension as a destination”. The word extension is hyperlinked, but the link is broken. Searching for extensions brings me to a page entitled Extensions | FreePBX. The information on this page is out of date, and attempting to make calls to external destinations using the Local/18005551212@outbound-allroutes style syntax (example provided by the page) says that I must “Enter a valid Dial string”.
Elsewhere on the site it’s discussed that this style is deprecated (for example, in this thread: http://www.freepbx.org/forum/solved-does-custom-extension-dial-feature-work ) and that I should instead use a different method, which also doesn’t work; the line is simply cut rather than the call going out.
The reason why I’m “angsting”, as you put it, is because I find outdated or incorrect documentation to be significantly worse than no documentation at all, and am very frustrated with my experiences with freePBX and the hours I’ve spent trying to learn how to configure it.
I always have problems with the self entitled who BMW everytime the spoon doesn’t fit their particular mouth.
As suggested please read all the documentation until apprehension kicks in.
Apparently , and I am guessing here, you might need a custom Extension which will dial SOMETECHNOLOGY/SOMETRUNK/YOURCELLPHONENUMBER for a virtual extension that is not local.
It is up to you to define SOMETRUNK using SOMETECHKNOWLOGY that can reach YOURCELLPHONENUMBER
it might look like dial SIP/all-outbound-routes/123456789, but it is you who designs the system not the system.
Well, see, if the spoon dissolves in liquid, and I’m trying to drink soup with it, that’s what I would consider a bug, rather than a feature.
I have read the documentation as suggested - note that I refer specifically to issues in the documentation - and apprehension kicked in some days ago when I realized that the documentation is inconsistent and in places entirely outdated.
The documentation is in the form of a wiki, that you put no effort into reducing any noticed discrepencies, as most who BMW about them without raising a finger to help do, pretty well guarantees entropy, you paid nothing for this software, you got a Ferrari, there is an instruction manual in the glovebox with an invite to correct percieved errors, where it belongs,and you did what to help the next BMW like yourself ? . . .
That was a bit hard to follow. Try this, under extensions, add a new extension and choose ‘Other (custom) device’ and in the field that says this device uses custom technology use this format:
Local/8435551212@outbound-allroutes
Replacing the numbers with the cellphone number of your end users. You can then use this extension as your call target elsewhere on the system, such as follow-me routing, etc, or if the user’s don’t have regular extensions, you can use it as their primary extension.
Yes. I want to route calls to cellphones. I don’t care if it’s over the PSTN or over IP; I need the system set up such that a call to the service results in an account manager’s cellphone ringing.
OK…There are a number of ways to do this.
But a few additional questions:
These are questions that need to be answered before you design your call flow.
BF
@dicko - I find it hilarious that you call this “got a Ferrari” when it’s more like “got a box of parts that claims to be a Ferrari, but the documentation is half in Swedish and half in Tagalag”. Followed, of course, by people arguing that if I don’t want to go learn Swedish and Tagalag in order to use the product, that’s my personal problem and not a problem with the product.
@reconwireless Last time I did that it gave me an error of “Please enter a valid Dial” or something along those lines, but I’ll try it again when I get home and have time to fiddle with this again.
Instead of pointing out everything wrong with the project why don’tyou simply tell us what you want it to do, what you have done and what’s not working. Don’t just tell Preston I tried that and it didn’t work. Tell us the exact dial string that failed and include the dialplan log trace from the time of the error.
Surely we know that the documentation needs improvement. Most everyone is willing to tell us how bad it is, few are willing to assist in editing or authoring content.
I think that using the resources we have toward development serves the widest audience. If you take the time to learn Asterisk FreePBX is “self documenting”
I do realize what is intuitively obvious to me might elude another.
There can by definition be no “bug” in the documentation apart from it is all buggy, it is free form, it is without doubt incomplete and aged. BUT if you note an error, please post it , right there where you read it. the bugs will then maybe eventually be swatted, does that make sense or will you passively wait for someone else to notice what you did already and post against what you have so far noticed, documented but refused to do anything about?
Just a thought . . .
You’re right; while it’s intuitively obvious to me that the current state of the documentation (where rather than, y’know, using the documentation, I have to trial-and-error against the system to learn how to use it) is literally worse than useless, that’s not obvious to anyone who’s emotionally invested in the project. However, I assure you, that is the case; inconsistent or incorrect documentation is much more frustrating than no documentation at all, and it is the former, rather than the latter, that you have.
I think this thread has pretty conclusively demonstrated to me that freePBX isn’t in usable state for anyone who isn’t already an expert in telephony in general and Asterisk in specific, but that the community believes that not to be the case.
Given that, and given the 20+ hours of work I’ve put into trying to get the system working, the migraines, and the half-hysterical “Why is it ringing… there’s an option for ringing, and it’s NOT CHECKED…” which got stares from everyone in the office, I think it’s best if I simply use Avaya, and everyone ignores this thread and goes on in blithe ignorance of how dense and unusable the system is for anyone not already an expert.
Goodbye.
Well, apart from all the many thousands of other non-experts who got it working successfully ovr the last few years, you might be right, it is way to complicated for some folks. But I suggest that really you leave us with many thinking that do just didn’t try hard enough, or just want someone else to do it for you, if so:-
http://www.freepbx.org/freepbx-official-paid-support
might be your simple recourse, but be aware, just like Avaya, it will cost you.