Author Topic: VoIP On the 6000  (Read 63225 times)

duffman

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VoIP On the 6000
« on: May 02, 2004, 02:02:47 am »
I tried LinPhone, no dice. Has anyone got it to work? Or know of any other VoIP apps for the 6000? Thanks!
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benkrembs

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« Reply #1 on: May 02, 2004, 09:27:52 am »
I also tried it a while ago and no luck.  But yeah.. VoIP on this thing would be AWESOME.  

Anybody out there talking on this thing?

coolass

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« Reply #2 on: May 02, 2004, 12:55:04 pm »
tkcphone is ok. Try the demo first.

duffman

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« Reply #3 on: June 02, 2004, 02:19:47 pm »
Coolass,

Did you notice any caveats installing tkc-phone? I got the demo and the support libraries but having some problems getting it to work still.

Thanks!
-Dustin
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tdt

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« Reply #4 on: June 03, 2004, 02:03:54 pm »
I\'ve also been trying to get this working with no luck. I\'m able to get the sip call set up and according to the ethereal traces the sip phone at the other end is sending RTP packets, so I think that the inbound direction is working. But nothing is ever sent from the zaurus side. I also get the following error messages when run at the shell prompt. i\'m not sure how to interpret them though. Does anyone have any idea about this? Terry.

!!!!!!!!!!!!!!!!!!!!!!!!!! Error: !!!!!!!!!!!!!!!!!!!!!!!
Data won\'t fit within the current RTP packet size
sent 54615 (3),received 55011 (3);read 529600 write 533440 need 528000 jitter 23
sent 55308 (3),received 55737 (3);read 536320 write 540480 need 528000 jitter 9
!!!!!!!!!!!!!!!!!!!!!!!!!! Error: !!!!!!!!!!!!!!!!!!!!!!!
Data won\'t fit within the current RTP packet size
!!!!!!!!!!!!!!!!!!!!!!!!!! Error: !!!!!!!!!!!!!!!!!!!!!!!
Data won\'t fit within the current RTP packet size
!!!!!!!!!!!!!!!!!!!!!!!!!! Error: !!!!!!!!!!!!!!!!!!!!!!!
Data won\'t fit within the current RTP packet size
sent 56166 (3),received 56496 (3);read 544640 write 547840 need 544000 jitter 16
!!!!!!!!!!!!!!!!!!!!!!!!!! Error: !!!!!!!!!!!!!!!!!!!!!!!

etc.

selfabuse

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VoIP On the 6000
« Reply #5 on: December 14, 2004, 06:26:45 pm »
Not to dig up a real old topic, but I got this working. It was a piece of cake actually. Grabbed a copy of KPhone/Pi from here, installed openssl from here, and away it went.

I have it connecting to an asterisk server back at my office, and it seems to work just fine. I just got the 6000 not an hour ago, and this is the first thing I tried

offroadgeek

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« Reply #6 on: December 19, 2004, 02:07:09 pm »
selfabuse - thanks for the post!  kphone/pi is sweet.. it works awesome on 6000.  

As soon as I got kpone/pi setup, I registered for accounts on the following VOIP networks:

Voice-Pulse
Nufone
Free World Dialup (FWD)

Here are some thoughts on the three:

FWD seems to be the best so far
- comprehensive website with nice features (call history, messages, etc.)
- great configuration instructions on website (even specific instructions for kphone!)
- offers multiple ways to test setup/connection (echo test, call back, and even a number to call (55555) to speak to a real person - kinda funny)
- no setup costs at all

Voice-Pulse  
- $10 non-refundable cost (adds $10 to your account for PSTN calls)
- decent site with limited configuration instructions
- I couldn't get kphone to register to voice-pulse network, even though I used the correct login/password
- I also tried the config on the x-lite softphone on my powerbook, still couldn't register  

Nufone
- works fine, but they have a very simplistic website and features.  Hardly anything for configuration instructions (except for the tiny bit you get in the confirmation email)
- was able to configure it and could receive calls via my ID@ipaddress, but couldn't get STUN setup (so that I could receive calls at offroadgeek@switch-2.nufone.net)

Using the 6000 as a VOIP phone is really nice.  It's totally comfortable to put the back of the zaurus to your ear and use it like a regular phone.  My test calls were very clear.

I wonder how many people with a 6000 will actually use it as a voip phone?  Also, if you want to do any reading on anything VOIP, check out voip-info.org

btw - I put my voip # in my signature, in case someone wants to do a test call
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selfabuse

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« Reply #7 on: December 19, 2004, 04:45:40 pm »
If you're in the mood to be super-geeky and you have a linux box at home, you can even pick up a pretty inexpensive PCI card, install Asterisk, and do VoIP on anywhere you've got a decent internet connection on your 6000 back to your landline at home.

After seeing me making VoIP calls on this, the president and vice president of my company have started shopping around for 6000s. We just started selling VoIP service, and VoIP PDAs would definatley be a cool thing to show off to prospective customers.

offroadgeek

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« Reply #8 on: December 19, 2004, 04:51:37 pm »
Quote
If you're in the mood to be super-geeky and you have a linux box at home, you can even pick up a pretty inexpensive PCI card, install Asterisk, and do VoIP on anywhere you've got a decent internet connection on your 6000 back to your landline at home.
I already have asterisk running on my of the ZUG servers in the house   though I don't have a pci card installed to connect it to my landline.  I'm still tinkering with asterisk... I'd like to have it configured to be a STUN server, so I can have an address like offroadgeek@zaurususergroup.com (or something like that), and be able to log into it from anywhere.

The nice thing about the VOIP services that I mentioned before, they offer real cheap international rates, in case I ever need to call the UK or japan... but I guess I hardly ever do that.  Then again, it would be cool to call my nephiew who's stationed in Japan.

I wonder if we should setup a VOIP ZUG server for our members to use?  Though I can't think of anything that we could offer over any of the free providers...
« Last Edit: December 19, 2004, 04:52:37 pm by offroadgeek »
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xjqian

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« Reply #9 on: December 20, 2004, 10:06:21 am »
FWD with kphone/pi_0.9.8 + openssl_0.9.7d works great for me. However, only tried 163(echo) and 55555 (supposed to be FWD welcome phone, reached a real person once). Most of the time 55555 gives me a "Forbidden" error, which, I guess, translates to "busy". calling 10000(supposed to be talking to FWD's founder) gives back "non-invite response".

Waiting for my girlfriend to set-up her FWD, so I can do a reality test. Thanks selfabuse for the heads up.
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cvmiller

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« Reply #10 on: December 24, 2004, 05:08:12 pm »
Hi All!

So you got me all exited about Kphone/pi on my SL-6000. Got it installed no problem. I thought I would test it out with the Kphone on my laptop (v 2.03 included with Mandrake 10.1).

Everything seems to run OK. But when I actually call my Zaurus (from the linux laptop) an I use the default Codec G711u, I get nothing but noice on BOTH machines! And if talk into the Zaurus mic, the noice on the laptop momentarily stops (while I am talking) and then instantly resumes.

If I switch to one of the other Codecs (GSM, iLBC) then I get silence, but I don't get any voice either.

Any thoughts on how to test this before I connect to FWD or PND? I'd like to see it work on my local network first.

Thanks,

Craig...
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offroadgeek

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« Reply #11 on: December 25, 2004, 03:21:48 am »
Quote
Any thoughts on how to test this before I connect to FWD or PND? I'd like to see it work on my local network first.
I don't know that you can do direct IP to IP calls with kphone/pi (I could be wrong).  AFAIK, kphone/pi is a SIP only softphone, and allows only SIP calls.

I've found FWD the easiest to setup with kphone/pi and they have great options for testing calls (echo test, time, live person testing).  check out this config doc for specific instructions for setting up kphone/pi with FWD using STUN (in case you're behind a firewall).

good luck
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xjqian

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« Reply #12 on: December 25, 2004, 03:42:36 am »
I concur with offroadgeek. SIP is an index service provided by FWD or others. Although direct IP to IP calls with kphone/pi may be possible, it should be a different configuration.

Have anybody tried FWD Peering? I'm mosted interested in Vonage and Packet8. How's the quality?

VoIP on 6K is great. Isn't that one of the major reasons 6K designed for?  
« Last Edit: December 25, 2004, 03:42:58 am by xjqian »
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offroadgeek

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« Reply #13 on: December 25, 2004, 11:54:58 am »
Quote
Have anybody tried FWD Peering? I'm mosted interested in Vonage and Packet8. How's the quality?

VoIP on 6K is great. Isn't that one of the major reasons 6K designed for? 
I tried the peering number for an 800 call on my calling card (to see if I can reach a regular land line), but the DTMF doesn't show the full 10 key num pad, so I couldn't dial a zero.  I couldn't figure out a way to resize the window.  (I submitted a bug report for this)

In my excitement to get this working, I tried setting up an inbound DID so that I could receive calls on my 6000.  I happen to be in a place where my cell doesn't work, but the cabin has dsl and wifi so I thought it would be a good time to really test this out.  So I first setup an asterisk server, and got a DID through Voicepulse Connect, but I couldn't get asterisk configured properly to register with the voicepulse connect service, blah blah blah... I just couldn't get it to work.

So I thought I'd try Vonage.  Keep in mind this is just three days ago.  I setup a new account with vonage and ordered the Premium Unlimited plan with the softphone add-on.  After it was setup (within hours) I tried to setup the service on my 6000, but found out that Vonage doesn't support SIP only based softphones    I even called Vonage technical support to verify this.  So I got a full refund (including setup costs).  Overall, it seems like a good company, but it just won't work with kphone on the 6000.

Next I thought I'd try the full Voicepulse service (not just Connect).  This time I called Voicepulse technical support and they confirmed that they do indeed support SIP based softphones.     They have a 30-day money back guarantee, so I figured I couldn't lose.  I setup an account (America Unlimited) through them a day and a half ago, and the account is "still pending verification".   Definitely not as quick as vonage.  

In the mean time, I'm still tinkering with setting up the asterisk server correctly.  This would definitely be the most inexpensive route to go, as there would be no monthy charge except for the Voicepulse Connect service (which is minimal).

I'll let you know if I ever have any luck with voicepulse.
« Last Edit: December 25, 2004, 11:55:34 am by offroadgeek »
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unblowupable

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« Reply #14 on: December 25, 2004, 11:36:07 pm »
Great work so far offroadgeek! I look forward to learning how you make out with asterisk/Voicepulse Connect. It does seem like the ideal solution, but in the meantime I may try IConnectHere.com's "Build Your Own Plan" for SIP users w/ FWD. Anyone have feedback about ICH? According to FWD they should work fine for PSTN calls, and the prices seem reasonable.

FWIW I was able to use peering through FWD to call toll free numbers (Just had to dial *18#########). Additionally, I was able to use a calling card to reach a regular land line. I ran into the DTMF/keypad problem initially but the AT&T card I used allowed voice recognition for PIN and phone # to call. The quality of the call was very good. So basically if you have a 6000, kphone/pi, FREE FWD account, broadband connection and a callling card that doesn't require DTMF for the PIN and phone #, you can make calls to anywhere! Of course you can always call US toll free #'s directly through FWD. A phone #, voice mail and all the services of ICH, Voicepulse, etc., would be nice, but this is a quick & dirty solution that works!
« Last Edit: December 25, 2004, 11:49:14 pm by unblowupable »