Hey guys,
What is the best free softphone avalible?
I have tried a few and each one has its ups and downs, but none have really made the cut?
Are there any ones that any of you have had in particular that have worked.
I want simple , problem free, easy to use and it must be good quality.
Thanks
I would suggest Zoiper zoiper.com/
I like this piece of software alot for its IAX2 support. Dealing with a NAT or several NAT’s is no fun, except in a lab environment. If you dont know much about the IAX2 protocol, Read up. Zoipers inclusion of the protocol makes it superior to other softphones. Zoiper does work very well as a piece of software.
Now if the service you wish to use doesnt offer IAX2 support, Zoiper is still a good choice for SIP. Its easy to setup and free.
Oh ok,
thats the one i am currently using just allot of my clients moan about bad quality audio and volume.
Theys ay skype works perfectly but their zoiper is very soft? when the volume is on full?
You do use the free CLASSIC version right?
And for iax2 do you just add the extention in freepbx and put the password in zoiper and it should work? Cause currently i am using SIP.
Would you rather use sip or IAX2 on internal network? Cause iax2 is easier to catch in routers for priorotization as it only uses one port???
I believe the last version of zoiper I used was communicator free. And as far as audio problems with it, I dont recall any issues with it. The problem was with windows sound mixer(the inputs and outputs). And as for skype… The devil is in the details. I have no love for that product. Thats a whole other discussion. The quick point is I learned how to use Asterisk to get away from skype and gain privacy and content control.
Now using SIP or IAX, 90% of my endpoints use SIP(all of them hardware). The other 10% use zoiper and connect using IAX. My endpoints are scattered all over the place. All of my trunks are IAX2, All my DIDs are IAX aswell. I have a VOIP lab at home, and I have found using a completely different subnet helps. It is a QoS strategy I find works very well. When i have a say in the way enpoints are layed out in an Intranet, I prefer the voip equipment be completely segregated from the rest of the data network all the way down to the WAN Gateway. But managing traffic through one port is alot easier than with SIP. So I guess what i mean to say is if your gonna use a softphone stay with one that supports IAX and use it.
Setting up an account for someone with IAX is really no different than with SIP. Username and password setup is the same really. There are a few things involving call tokens, Dealing with it can be tricky. Its all worth the effort though.
Thanks,
i like the diff subnet idea?
So if ur pcs are running on 192.168.1.0/24
u make your PBX 192.168.1.2/25 ???
or like a diff ip range? 192.168.0.1 ?
I suppose mine is a strange setup, but here goes.
WAN -> ROUTER-1 -> ROUTER-2A & ROUTER2B
ROUTER-1 Has explicit control over QoS, hands out static IPs.(e.g. 192.168.0.1/24)
ROUTER-2A Is connected to a switch, Uses QoS for its sub-net, Computers connect to switch, Data use only. Hands out IPs with DHCP.(e.g. 192.168.1.1/24)
ROUTER-2B Has Qos features, Asterisk Connects to it, FXS devices connect to asterisk here aswell. Hands out IPs with DHCP.(e.g. 192.168.2.1/24)
ROUTER-1 Offers the best quality connection to asterisk all the time, Trunks and DIDs come first, Than the FXS devices.
Since each layer has its own control, It is fairly easy to manage. You can get into things like VLAN tagging and such, But I personally havent really played with that much. It offers a very useful way to get QoS on your incoming and outgoing packets on a large network.
I get to play in a lab most of the time, Most dont have this luxury. Can you describe your networking environment a little?
well i tried a IAX and it didnt work???
I created a iax extention on freepbx and then put the username and password in zoiper and it wouldnt register???
that exact zoiper works fine with SIP. What could be wrong?
Well the setup i have is a temporay setup. I am testing.
i an starting a WISP soon and i want to provide VoIP to my customers.
I i will have one router to which the PBX plugs in and then the wireless bridge which goes to my tower with 3 sectors on it?
So i want to setup the most ideal reliable (simple) VoIP network that will be able to provide 50 users with phones>
Im sarting out with 5 but i want to ahve 50 in time?
So any advice and suggestions is great.
And with your other subnet thing. how do the computers connect to the PBX then? if they are not on that subnet?
Well I’m not to sure what to say about freePBX, I started out my adventures with VOIP and PBX’s with freePBX, My second choice had been askozia. I eventually gave up on them because of the gui, and what felt like a lack of simple controls. I wouldnt really be the one to tell you how to setup an IAX account.
I think your into a fun and ambitious project, I think you should find a hardware solution for your problem, Like an ATA device.
They offer so much more control and keep service issues to a minimum as a failing computer doesnt remove access to a phone call or even PBX features. later today im going to try adding a few iax users to my server(internally), As my current users make there access via the WAN.
I do not want users to have the ability to directly access my server equipment. Once I write all the configuration files needed I let my machines do there work. I must state that this is not the kind of work I do to pay my bills. I actually work as a laborer, So I really only do this for the fun of it, I also take it as seriously as possible.
For instance,
All of my servers get tailored kernels suited for their particular tasks.
I build as much from source code as possible. This can get tedious.(worth it though)
I also abuse my servers every couple of weeks from random locations with random port scanning attacks and such, and read the logs at every hop from inside my network.
Ive considered letting another person abuse the equipment, to see if i can track down those problems.(I should have done this a long time ago)
My main asterisk server is not located on my premises, I have a satellite server here at home ,actually its one of my old projects which is a hacked gamecube, I Love Power PCs. This is where I learned to build Kernels. Embedded systems are tough at first but so useful. I also have an affection for the Debian flavour of linux.
Can you possibly tell me what the CLI reports when your IAX client tries to connect. There is more than one way to skin a cat and linux has proven this to me. Im pretty sure a solution can be found. I would love to help further. Feel free to E-mail me directly.
I figured it out.
I had to type “no” by calltokens…
And that seemed to work. To me it seemed that IAX2 was better quality than SIP over the internet?
I was testing from work. the PBX is at home.
I’m a pretty big fan of the IAX2 protocol myself. The real neat stuff is the trunking you can do between servers. Can save alot on IP overhead. Be careful with the call token support, not treating it right can leave you vulnerable.
What exactly is call toaken?
How can i make Zoiper send a call token instead of me saying “no” under extention.
And what codec would you recomend?
i need one for 2 scenarios:
A:
good quality
B:
speed ( so i can have many calls at once)
I have not spent a whole lot of time with the calltoken support, I do know that it is an extra precaution used to thwart call number exhaustion. I dont believe that call token support really has anything to do with the client end, I’m pretty sure(dont hold me to this) that it involves how asterisk authenticates the clients priveledge to make and recieve calls, I really should know more about this and how it works.
As far as codecs are concerned this depends on your needs. I like G.729. G.729 has a license cost involved with it. The gsm codec is a good choice/performer. ILBC supports PLC(Packet loss concealment)which is useful if a client is subject to minor packet losses. G.711 is just to bandwidth expensive a codec to use, sounds great though. If money is no object and you want good performance G.729.
The lower the bitrate, the higher the demand on the CPU. From the asterisk console run this command “core show translation” and you will see the comparison between codecs and thier costs.
Ok,
well i have 5 phone clients over a longrange wireless bridge:
So gsm or the one with packet loss protection?
3 on zoiper,
and then one Grandstream VoIP sip phone and one Grandstream ATA Device.
So from your previous post i would say i should use GSM for these devices as it will use the least amount of data?
SO you act have to pay for the one you like?
How? do you just buy the software and laod it on your pbx or do u pay for each phone?
and u-law ? i see asterisk uses that as default?
I’d say your choice is pretty clear, GSM. Most Softphones support it. I cant really recall using any hardware endpoints that have supported gsm though. G.729 is a really sweet codec, In my opinion. It sounds really good for the bitrate. Getting licenses for asterisk isnt to difficult, Digium can provide em to ya. Most hardware endpoints come with many licensed codecs, The problem is not have said licensed codec for asterisk.
Endpoint A has G.729 -|Asterisk|- Endpoint B has G.729, asterisk can pass the media right along. No license required.
Endpoint A has G.729 -|Asterisk|- Endpoint B has GSM , Unless asterisk has G.729 license transcoding is an issue.
Endpoint A has G.729 -|Asterisk-Handling the media|- Endpoint B has G.729, Three license’s are required here. In this example asterisk could moniter the call media.
G.729 is the codec made for PBX’s, Look at a numberpad 729 spells PBX.Ulaw,Alaw or G.711are the 64kb/sec telecom standard. It does sound good but bandwidth expensive. Its one to use inside of an intranet.
Oh ok i see. Thanks, That makes more sense.
So how much are these licences? for 729?
Around ten bucks a piece, this depends i suppose on several factors like exchange rates and such. I have heard they can be purchased from a licensing pool. I am interested in finding out more myself. When I do i will post.
If you dont mind me asking, What kind of hardware do you use for your WISP project, and Asterisk?
I’m really interested in this, Sounds like fun to me!
Ok with asterisk,
I just have a intel board with a 2ghz celeron and 1gb ram with 2*80gb hard drives in raid 0 mode, 4 port Diguim FXO card.
Then for the wisp i use wireless equpment called Ubiquity http://www.ubnt.com great stuff, easy to setup and works like a dam BOMB.
How would i go about adding faxes to my system?
i currently have 4 pstn lines coming into my PBX with a 4 port TDM card. How could i send / recive faxes through a (unbusy) port
Or set it up so that it sends as a fax to email? then i can just get a free fax to email number in which it forwards faxes to the apporpriate email address.
Thanx for the info. Now as for faxing, I cant really help much with that.
I have only tried sending fax using SIP and IAX2. My experience was terrible with essentially only a one way experience. I could send but never recieve.
I hope someone here can help you more.
Well how hard is it to send faxes? Cause i can just use a fax to email system to recieve?
How do you fax? Do you use fax machine or do you click print on the computer and it faxes?