[DEAD AND BURRIED] WM6 SIP Config Tool - [V2.0.1 Out Now]

I think that error (parsing) only appears when you try to load your config for the FIRST time. And you had Voip already working. Now if you fill out all fields manually and save them using this tool and then you will be able to load them w/o any problems.

bogdatov said:

I think that error (parsing) only appears when you try to load your config for the FIRST time. And you had Voip already working. Now if you fill out all fields manually and save them using this tool and then you will be able to load them w/o any problems.

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Congratulations, Bogdatov, you're right ! If I fill up the same data already provisioned on my device, but now using this application, and save them, then I'm able to restore the same data through it. Problem solved !

So, Shaun, please take a look at it. Although Bogdatov has found a workaround, I suppose no application should require the same data population twice, don't you think ?

Thank you very much for your hint, which has gone right to the point !
Regards,

Carlos

I've just had some fun trying to play with VoIP on my hermes, i have got everything working great and now i find this tool

I try to load config from device, and the tool is crashing

here is my voip config (_setup.xml)


I've created my cab and got my device working fine, i'm only posted here to try and help improve this tool. When i try and load config from device, i get


If i select details, the exception is an ArgumentOutOfRangeException

edit:

I've just noticed a couple of other have already reported this, but you asked if they had already provisioned VoIP, well i can confirm I have (and my settings are above)

Also for info, i am on a HERM300 (T-Mobile Vario II) running the latest LVSW WM6 full rom

Proposition

Hi Shaun3, please take a look at my entry here: http://xdaforums.com/showpost.php?p=1237964&postcount=539.
I would be glad if you were interested in this.

nbedford said:

I've just had some fun trying to play with VoIP on my hermes, i have got everything working great and now i find this tool

here is my voip config (_setup.xml)

Code:

<wap-provisioningdoc>
<characteristic type="VoIP">
<parm name="SIPSettings" 
   value="<provision key='1232ab01' name='Sipgate'>=

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Did you manage to make sipgate work with WM6's VoIP client? :-o I've tried to set up my sipgate account today, but I only get 'No Service' message promptly after 'Searching...'. I was using SIP Config Tool to set up account settings.

Shaun, thanks again for the great program. I have a request though.

For aesthetics, is there any way we can make an icon for this program so it looks decent in the Programs menu. I'm also making a request for SmartSKey. This isn't a necessity, but certainly improves the experience.

Shaun33 said:

I was going to put an icon on it but it make the size of the app 300k
now thats a big jump just for a pretty picture.

I will look into it some more for the next versions.

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If you already have one, would you pass it along? I'd like to put it in my ROM.

Shaun33 said:

When i find it again. I am going to release a new version soon; so rather than putting it in the rom i would be happier if you just linked to this thread.

Im hoping to have the dialplan editor out within a week or so.

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No worries...I'll give credit where it's due for sure.

Shaun33 said:

Have updated and here is the new release.
Its in the second post. Or ya could just click the link.
Verison 2.0.1

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Tested and working, thanks a lot Shaun33

Will you consider adapting your application to work with Smartphones as well as PocketPC devices? I can load your cab file, but it errors out with an exception about unsupported device. I am able to make VOIP work just fine with my Dash, but setting it up was a pain and my dialplan is still in shambles. I would love a UI tool to do some of the heavy lifting and it would be greatly appreciated if you could adapt your program to the smartphone platform.

Thanks

hi guys
i'm realy a newby with voip and couldn't configur it with this cab i have an account in voipstunt ,can somebody help me to configur correctly please? i have giust configur it in this way but dosen't work
1- despription :voipstunt.com
2- sip server : sip.voipstunt.com
3-username : xxxxx
4-password :xxxxx
and after enabled sip over 3g/Gsm ,maby it dosen't work because i try it only in gprs ? it works only in wifi connections ?

Shaun33 said:

Dialplans is going to be in the next release.
So just hang in there and it will come out soon.

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Unfortunately, my problem is not about dialplan.
My problem is about strange behavior of RTC in my Uni during establishing a call.

But first thing is the first:
I started with creating the _setup.xml as well as ipdialplan.xml according to description given by eluth. I checked my dialplan using http://www.roblocher.com/technotes/regexp.aspx service and after some corrections it seems to be OK.
Then after putting it in to my Uni, I started to make some tests.

I have to say that I run my own commercial VoIP network based on servers based in Brussels/Belgium. I use PortaOne software named PortaSwitch to run the system. The software is based on Sip Express (SER), but commercially redeveloped by PortaOne from Canada.

Normally I use in my work many hardware and software VoIP clients. In particular I use X-PDA and AGEPhone softphones on my Uni. I usually use them through 3G, as my network gives pretty good connection in main cities. Sometime, when I have WiFi and 3G is not in the place, I switch to WiFi.

Now coming to the point, after loading the prepared voip setup in to my Uni I have found strange situation. I have to say that later I have used Your SIP Config Tool to recreate the SIP setup in my Uni, but this did not change anything.

The situation is as follows:
First I have made tests through 3G.
1.
When I started the settings, Uni has registered in my VoIP server and properly showed that the "EuroTELEFON" service is selected.
2.
When I placed a call from other VoIP phone to my Uni, it was marvelous. The Uni showed info page about the coming call (with number showed and my name taken from my contacts). I picked up the call and audio was at least good in both directions. I checked the info about the call on the other phone and found that the call was on G711a/G711u codec (for each direction).

I could place such calls as many as I wanted to and all were in the same level of quality.
3.
Then I tried to initiate the call from my Uni. And here the problem came. The other phone did not even started to ring. My uni showed an info that the service is not available, then the call ended. I checked logs in the server and found that the attempt of the cal was not even registered.
4.
When again I wanted to call my uni from other phone, it rang but after pushing the pickup button the Phone Program got stacked. The call info page did not disappear, the call was not answered (the other phone was giving the dialing tone all the time) and the only think I could do was to reset the uni.

Then next test I have done using WiFi.
5.
It registered as always in my voip server and showed that the EuroTELEFON service is selected.
6.
Then I tried to call my uni from other phone. This test on 3G was passed, but now on WiFi the result was exactly like with similar call on 3G but after the call from uni (step 4). In addition I have found that all my home network (connected by ADSL with IP changed after each reset of the router) has bean cut off by my voip server. This is standard behavior of my voip server if it founds that somebody is trying to brake in or doing some harm to the server.
7.
I had to reste my router to get new IP and have done the test with call initiated by uni. Result exactly like in step 6 and the other phone did not even ring.

Then I have changed the settings of my uni to FWD account. And strange result. All calls in both directions were OK. The quality of the call was not as good as in the test in step 2. but at least Uni was calling and receiving calls.

Now my problem is that I would like you guys to help me to solve the problem. That is way was my proposition. I am ready to give you free accounts and am ready to supply some logs taken from the server. I suppose the problem is with the way RTC is making the process of digestion of its login and then establishing a call. I am not sure, but I remember that once I,ve tested some softphone for WinXP based on RTC project and got similar results. I think I also could ask people from PortaOne on their forum to help us.

Sorry for that long input in the forum but still hoping that somebody will be interested in helping.

Dash with T-Mo WM6

I have installed 2.0.1 and tried to run it. I get a NotSupportedException Error at Microsoft.AGL.Common>MISC.HandleAr()

Any ideas?

Dave P