sundance-communications.com/forum Forums VOIP Phones & IP PBX VoIP & Cloud Phones, Hosted PBX, General IP Phones Sunrocket VOIP and phone system routing

Hello everyone,

I have a SunRocket VOIP phone service of which plugs into a gizmo and then into my phone and it has been working great. There are some features that aren't implemented of which I will need additional hardware for. Technically they say that SunRocket is capable of going into your entire phone system if need be, but I only have it going from the gizmo straight into one phone.

I want a system where the Sunrocket plugs into "it" and ultimately "it" has its own routing features. For example, it will automatically pick up and go on with a recording "Thank you for calling the Jones', for Jenny press 1, for Jerald press 2, for John press 3" of which if someone presses 1, then Jenny's phone in her room will ring, or it will reroute to her cell phone.

I'm not very familiar with phone systems, but I do know that this sort of equipment does exist, whether it is end-user consumer grade or not, I think I could figure it out, I just sort of need a push in the right direction. Can anyone point me in the right direction on what would be the best route for something like this?

Feel free to ask questions to get a better understanding of my situation as I would really like a solution. Thanks!

This is only half a VoIP question.

First, your 'Gizmo' is called ATA- Analog Telephony Adapter. The network side connects back to SunRocket using a protocol called SIP (Session Initiation Protocol). The analog port provide POTS (plain old telephone service) service to a phone or other gadgets.

The next thing you want is a voice menu (called an IVR menu (interactive voice response) or auto attendant).


There are two ways to do what you want.
The first, and possibly easier way- with a normal PBX. There are many inexpensive PBX systems that will work quite nicely. You will need one with auto attendant capability that can handle as many lines and extensions as you have. Many can support a larger number of extensions or lines but require additional hardware (expander modules) to do it, so make sure you are buying what you need and have the right modules. This will cost at least several hundred dollars. It will also provide you with basic local call routing and other features (you can dial each other by extension, transfer, intercom, conference, etc). You can also expand the system to include other doodads like door phone intercoms and music on hold.

The other way (far moreso my area of expertise) is with an Soft-PBX, AKA Asterisk. Asterisk is an open-source (and thus free) software based PBX system that runs on Linux.
If you have no idea what I'm talking about and you don't want to learn and/or you don't have and can't get a spare low-end computer to do this with (pentium2 or better is recommended), then Asterisk probably isn't for you. If you want it to Just Work and aren't interested in learning about things, ignore all this and buy a small PBX. You might try Trixbox, which is a quick way to install asterisk with a web based interface onto a spare computer.
Asterisk (when configured) will do all the same stuff as the standard PBX, and also much more if you want it to. Asterisk is infinitely flexible and very cool. However it may be more than you want.
If you go this route, you will need some way of attaching to your SunRocket service and to your house phones. The best way is to have Asterisk talk directly to SunRocket servers via SIP, without using the ATA at all. From what I can tell, SunRocket does not promote or support this, but they don't actively prevent it either. Many VoIP providers (vonage) actively prevent this both through their terms of service and by locking the ATA. You can also install an interface card in your Asterisk server that will use the SunRocket line.
As for your phones, there are three ways of doing this. The cheapest is ATAs, you can use gadgets like your SunRocket ATA to connect to different phones in your house. You will need a few ATAs (because you want each phone to have its own ATA port, so extensions can dial each other). You then connect normal analog phones.
The next is with interface cards. These go in your Asterisk server and provide ports for your phones, just like ATAs do.
The last (and best in my opinion) is IP phones. IP phones are VoIP telephones that connect directly to your Ethernet network. If you don't have Cat5 wiring in your house, this may not be for you. IP phones are a better option though because they have physical buttons for features like transfer/conference/etc, instead of having to flash and use star codes.
I will stop myself there because I could go on for hours about VoIP stuff. If you are interested in Asterisk as an option, I will happily provide any info you need.


Overall- I think Asterisk is a very attractive option if you have (or are willing to install) Cat5 Ethernet wiring and IP phones, and are willing to learn some stuff about VoIP and Asterisk. If you don't have or don't want Cat5 (only have plain telephone wiring) and/or you want something really simple which Just Works, then a standard PBX will be a better option (because those phones only use 1 or 2 pairs of wire and wire in directly without any network gear).

Hope that helps!

IronHelix - Thank you so much for the information. I just so happen to have a low end (P3) system laying around and I do have experience with linux. So I believe the Asterisk way is the best way for me, and probably the most cost effective (cheapest) and most powerful from what you have said.

Is there a tutorial or something that is extremely detailed on exactly what you outlined for Asterisk as a solution for what I would like done?

In the meantime IronHelix, i'm going to be visiting asterisk.org which looks like the website of the solution you are recommending. Any tips/tricks that you can come up with at the top of your head would be greatly appreciated.

First, sorry for this long post. I can be very, er, verbose when i am talking about VoIP or Asterisk


There is a book, asterisk: the future of telephony. The dead-tree version is published by O'Rilley and should be available in most bookstores; or you can get it online (legally) for free at https://www.asteriskdocs.org/modules/tinycontent/index.php?id=11
Read the section on dialplan design (extensions.conf), once you understand the dialplan you can do anything. It's here that 90% of asterisk's intelligence lies.

Yes, asterisk.org is the official website. You can also find a ton of info at the Wiki, https://www.voip-info.org . Lots of asterisk-related info there.

For IP phones, I'd suggest buy yourself one or two Grandstream (BT200 ($50) or GXP2000 ($90) phones (not BT1xx if you can avoid it). These phones are relatively simple and many Asterisk users learned about SIP on GS hardware. You want the BT200 and not BT100 (if you get the cheap one) because the BT200 supports auto hangup and Intercom (auto hangup = speakerphone turns itself off when call ends, intercom = voice paging). GXP2000 is slightly higher end and supports this too. GXP2000 also supports power over ethernet (802.3af) so if you buy a powered switch you can juice them that way, makes things simpler. (netgear makes a nice 24 port switch (16 ports are powered, 16 unpowered, 2 gigabit uplink and minigbic ports); CDW.com's having a sale, it and a Wifi AP for $300)... model is FS726TP as i recall.
Also check out AAstra and SNOM IP phones- they are built like tanks and a joy to use. I use a SNOM 360 on my desk. SNOM and AAstra both make PoE compatible models. If you want cordless, Linksys makes a few decent ones (WIP300/WIP330). These phones speak SIP and connect via WiFi to Asterisk. The 330 has windows CE and you can check your email from it . It costs like $400 tho
Lastly, you can use a computer as an IP phone- using a 'softphone' program like X-Lite or SJphone (both free) you can connect to Asterisk and use your computer headset (or bluetooth) to talk. These are not a good replacement for real phones, however they can be quite convenient for making some calls from the PC. You can also get softphones for a PDA if it has WiFi.

As for your sunrocket service, you'd definately want to connect to it via SIP, not thru the gizmo. I looked and didn't find much info about how to use them with *. There is a post on the sunrocketforum site that basically says that if you want to BYOD (bring your own device), they don't support it and wont help you do it, but if you manage to make it work they won't stop you either. However from searching, it seem that they really, really won't help you- you have to (somehow) figure out the admin password for the 'gizmo' and from there you can apparently (sometimes) get the right info to make Asterisk work.
You really want to get it working directly via SIP if you can. Asterisk will deal with a SIP channel much better than with the analog line, getting call waiting to work in any useful capacity on a POTS line with asterisk is somewhat difficult. With SIP, it works flawlessly. Actually with SIP, most providers give you two channels (for 3way calling) so you can have two people on the phone at once.
You could also try porting your number to another service that supports BYOD, ie broadvoice, viatalk, etc. However you may not want to do this if you have one of the $200/year plans. Sunrocket DOES support porting out their numbers if you go that route however.


If you don't go with IP phones, and/or to connect to the gizmo or analog lines:
First, FXO and FXS. An FXO port connects to a Line and connects * to the phone service provided by this line. An FXS port provides dialtone and phone service to phone(s). An ATA usually has two FXS ports. If you get these mixed up, remember that FXS Serves.
There are two main types of analog cards you can get. There's the Digium TDM400 series (PCI card, up to 4 ports/card, each port is enabled as FXS or FXO by adding a module to the card. FXS modules are green, FXO modules are red.) There's also the Sangoma A200 'Remora' series (PCI based system. One PCI card connects to one or more 'Remora' cards via a backplane, each providing 4 analog ports. Remora cards take modules, each module enables 2 ports as FXO or FXS. Same coloring). These cards are a bit expensive from a home user perspective at around $70-90/port. If you want cheap FXO cards, you can get an X100 clone card on ebay or from a few places. The X100 is a discontinued Digium product which is basically a rebadged voice modem of a particular type. Sometimes they work, sometimes they cause difficult-to-solve echo problems, and sometimes they are the cause of a number of odd issues. They are cheap, at around $10/each.


I'm gonna ramble back to dialplan design for a sec to give you a basic idea of how it works. In asterisk, you have:
[context]
exten => extension,priority,application(argguments)

each context is a group of extensions. Every phone/channel/call is 'in' a context, and can only dial extensions in its context, and extensions in contexts included to its context, and so on (you can include one context in another). The extension is a number or number pattern that can be dialed. priority is what order it happens in. Application is what happens, and arguments are passed to the application.

For example, say you find your sip settings for a provider, and put them in sip.conf, naming the link [provider], putting them in context [incoming]. Say you also have five IP phones, named 10-15 also defined in sip.conf, them in the context [phones]. Your basic extensions.conf might look like this:


[incoming]
include => extensions
include => check-voicemail
exten => s,1,Answer() ; ; is the comment char. S is the 'start' exten. Answer picks up and needs no args.
exten => s,2,Background(welcome) ; s,2 is run after s,1 finishes (thats priority at work. Background plays an audio file, listening for DTMF.
exten => s,3,WaitExten(5) ; WaitExten waits while listening for DTMF. 5 is number of seconds
exten => s,4,Dial(SIP/10&SIP/12&SIP/13,20,r) ; Dials sip phones 10, 12 and 13 if they don't push anything else. dials for 20 seconds, plays ringing to caller. If no answer in 20sec we move on.
exten => s,5,VoiceMail(10) ; leaves voicemail for mailbox 10

[extensions]
exten => _XX,1,Dial(SIP/${EXTEN},20,r) ; matches any two digit number. ${EXTEN} is the number that was dialed. This lets you define 10-19 in one line insteadl of 10
exten => _XX,2,VoiceMail(${EXTEN}) ; if nobody answers in 20sec go to their VM.

[phones]
include => extensions
include => dial-out
include => check-voicemail

[dial-out]
exten => _1NXXNXXXXXX,1,Dial(SIP/provider/${EXTEN}) ; 11 digit numbers get passed straight to the sip provider.
exten => _NXXNXXXXXX,1,Goto(1${EXTEN},1) ; if you dial 10 digit. It puts the one in front and 'goes to' the resulting exten priority 1, which matches the first line.
exten => _NXXXXXX,1,Goto(1999${EXTEN},1) ; same as above. replace 999 with your area code.

[check-voicemail]
exten => 8500,1,VoiceMailMain() ; voicemailmain lets you check your VM.


The result:
when a call comes in from the provider, it starts at s,1 in [incoming]. It then executes the s priorities in order, which will: Answer the call; play a sound file, wait 5 seconds, dial three phones, and go to voicemail. If the caller types in an extension number during the welcome greeting or the 5 second delay, he will go straight to them (in [extensions], because it was included to incoming). If internal phones call each other by dialing two digits, it rings their destination and goes to voicemail. And if internal phones want to dial out (7 10 or 11 digit), it sends that out through the provider as a 11 digit number (dial-out context is included in phones). You can also check your VM by dialing 8500 from either inside or outside.
If you get more lines for your kids, you can put them in different contexts. A different context might have it's s,1 be just goto(extensions,13,1) which would send that call straight to exten 13, then to voicemail (as above).
Hopefully by now if you're still awake you are realizing how flexible this system is. I've shown you a few basic applications (dial voicemail voicemailmain goto answer), in reality there are over 160 applications that come with Asterisk. And if you don't like those, you can use AGI (Asterisk Gateway Interface), an API that lets you script out call flows in PHP, Perl, or just about any other language.

Also as an aside- be very careful with includes. Careless use of contexts and includes can give random people access to billable services. For example, if you include dial-out in incoming, then any caller can punch a 11 digit number as their 'extension' and the call will go through, on your dime!


Lastly, there is one thing (i've used Asterisk for a few years and I've only found one thing) that Asterisk cannot do: Shared Line Appearance. This is where an analog line gets a softkey on the phone, which lights up when it is in use, and pushing that button while lit conferences you in with the others already talking. This can be useful in a home environment. However, it's scheduled to be included in Asterisk 1.4, which should hit beta before the end of the month. (current version is 1.2.12 as i recall). As you may have imagined, Asterisk is named after the wildcard character, because it can talk to almost any type of phone/line/system, do anything with the call, and pass it off to any other phone/line/system.
As an example, I recently used Asterisk to replace the voicemail system on an outdated analog PBX. I posted a full writeup if interested:
https://www.voip-info.org/wiki/view/Asterisk-Partner+ACS+for+Voicemail


Anyway, hope that all helps, sorry for the length!

IronHelix - I want to let you know that you are awesome and your knowledge and williness to help others is rare.

I'll read over this stuff several times, read through some of the free PDF book and will post back here in this thread. I'm currently downloading TrixBox and Vmwarez to do some testing. I believe i'll install vmwarez first and once I get the hang of it and believe that this will work with my application, Trixbox for an actual linux dedicated box!

Glad I could help

Also one thing to keep in mind- Trixbox is NOT asterisk. Well it is sort of, but not really. Trixbox is to Asterisk as MacOSX is to BSD- it's a candy coated GUI shell. (except you still have to SSH trixbox to do certain things). The main thing I don't like about trixbox is that all the features create a LOT of complication. On * (plain), a call coming in, listening to a menu, making a selection and getting sent to a user generates 5-10 lines of console/log output. On Trixbox, you have a page of output by the time the menu even starts playing. This is because of all the various conditional things there are that must be processed (is the call going to be recorded? is this line forwarded to another line? is privacy manager turned on? Should this line ring another group? etc etc etc)... makes debugging trixbox far harder as there is a ton more output to muck through. If you are good with *nix, I'd recommend install CentOS (or your favorite distro) on the P3 box and start going from there, you'll probably find it easier. At the very least, it's somethign to think about. Good luck with your project!

I obviously have a lot of reading and learning ahead of me. I'll keep this thread updated though with my status.

So what you are saying is * is better than Trixbox is keeping it simple. With *, you just download and install CentOS, then install * from there, and then do a little editing and programming right and then testing!!!

Also what CentOS version would I need? I went to centos.org and noticed that there is 4, 3 and 2. Within there is SRPMS/alpha/i386/ia64/ s390/ s390x/ x86_64/

I assume i386.

the 'little editing' takes a few hours the first time, just be prepared

you want the latest version of centos, which is 4.4 as i recall. You would want the ISO image, which you burn to a CD to make an install disc. You can find some at the mirror here:
https://mirrors.easynews.com//linux/centos/4.4/isos/i386/
the other options (ia64 s390 etc) are for different CPU architectures. ia64 is Intel Itanium 64bit, x86_64 is AMD's 64bit variant for Athlon 64 / Athlon 64 X2 / Athlon FX series CPUs (which has been adopted by Intel with many newer processors).

For your P3, i386 is the right one . Link above, download, burn.

Also for the record, CentOS is very similar to Red Hat. A while ago, Red Hat decided to start charging for their OS. The 'free' version lived on inside RedHat as the Fedora Project (Asterisk works great under Fedora too). However because of the GPL license (it explicitly permits redistribution), it is legal for a group to take the RedHat enterprise linux software and redistribute it, as long as it isn't CALLED 'red hat' (the name redhat is a trademark). Thus, CentOS- it's redhat enterprise linux, minus the name redhat and a few logos.