[GUIDE] How to configure VoIP/SIP client in WM6.

butt · Mar 29, 2007

cabarc

Not wanting to appear to dence but i'll be buggered if i can get the cabarc command to create a cabfile my uni will recognise grabbed the download from the M$ site that has the cabarc.exe in it unziped it copied the _setup.xml file in to the same dir cut and pasted the command in the first post to create the cab as needed but it will not for love nor money install on my damn fone any ideas would be appreciated.

edeplano · Mar 29, 2007

butt said:
Not wanting to appear to dence but i'll be buggered if i can get the cabarc command to create a cabfile my uni will recognise grabbed the download from the M$ site that has the cabarc.exe in it unziped it copied the _setup.xml file in to the same dir cut and pasted the command in the first post to create the cab as needed but it will not for love nor money install on my damn fone any ideas would be appreciated.

Dont use a rightclick CABvia Activesync type command, you need to copy the cab file on a SD card and run it from your device

wratran · Mar 29, 2007

does this work with 3G service...i know it mentioned that it does not work with edge

johngillespie · Mar 29, 2007

Cynagen said:
Now that we have VoIP on our lovely little Wizards (aptly named I might add), is there any way to set up a SIP server say on a linux box that'll convert your home phone into VoIP? I know this is a little off-topic, but I work from home, and recently had to get a home-phone because I got charged with $800 in overages, so I'm hurting to find a way to null and void my cellphone's GSM link while in the house and have it use my home line for all my local calls, and keep my cell for long distance and just the random incoming call.

You need to buy a SIPURA unit to convert your analog home phone line to a SIP interface. Then you'll need to install an Asterisk SIP server.

have fun

Cynagen · Mar 29, 2007

johngillespie said:
You need to buy a SIPURA unit to convert your analog home phone line to a SIP interface. Then you'll need to install an Asterisk SIP server.

have fun

Screw that... analog modem FTW. Good idea on the * though.

Golota · Mar 29, 2007

Hi,

I have tried cabarc and created the cab file, but when I install, my Wizard says: cab file installation was unsuccessful. I tried the original txt file and the result is the same. Any thoughts?

Golfman · Mar 29, 2007

You can do this in several ways...I'm responding here as its SIP related, but we might want to a start a new topic to prevent cluttering this thread.

My focus here is for a working solution with Asterisk - note, Asterisk is not just a SIP proxy its a full PBX, it will handle TDM, SIP, SCCP, AIX and MGCP. Best way of running Asterisk for a novice is probably Asterisk@Home (its based on CentOs I believe). So it depends on what you want to do - see definition of FXO vs FXS.

1. Use an external ATA that converts POTS -> RJ45. Most of these are "locked" to a service provider such as Vonage. Some are available unlocked. They typically come with 2xFXS (provides dialtone versus FXO that expects a dialtone)

http://www.voipsupply.com/index.php?cPath=96_118

These products are aimed at plugging a convention phone into them and connecting direct to a VoIP service provider - in this case Asterisk (if the unit is unlocked)

2. If you simply want Asterisk to answer a POTS line, a TDM100P card (or clone) will do the trick. Its a glorified Intel V90 modem that acts as a single port FXO was was re-badged. Its no longer sold by Digium but range in price on eBay from $16-$30 depending on how "cloned" it is.

3. Preferred solution (also the most expensieve) is to use a TDM400P. You can then populate it with upto 4 x FXS or FXO modules to suite your needs. Expect to pay $140 - $350 depending on how many FXS/FXO's you add.

Cynagen said:
Now that we have VoIP on our lovely little Wizards (aptly named I might add), is there any way to set up a SIP server say on a linux box that'll convert your home phone into VoIP? I know this is a little off-topic, but I work from home, and recently had to get a home-phone because I got charged with $800 in overages, so I'm hurting to find a way to null and void my cellphone's GSM link while in the house and have it use my home line for all my local calls, and keep my cell for long distance and just the random incoming call.

Sleuth255 · Mar 29, 2007

We need the Holy Grail here though: VoIP over the 3G connection. I thought eliminating "Cell" from restrictions in ipdialplan.xml would do it but no joy

papamopps · Mar 29, 2007

Sleuth255 said:
bummer.... while calls transfer to my BT headset, there's no sound at all.... This will take a bit of troubleshooting I'm afraid.

Try using bttoogle.exe then you can manually switch every sound to headset - even music. But just mono.

BTW I have got still the problem with external speaker like others. Can someone who has it working on internal speaker upload their registry including VoIP settings to compare to us??

edeplano · Mar 29, 2007

Sleuth255 said:
We need the Holy Grail here though: VoIP over the 3G connection. I thought eliminating "Cell" from restrictions in ipdialplan.xml would do it but no joy

I tried using sjphone with orange 3g in london and the quality was no good, it has to do with latency I think and the latency on umts is not good enough although bandwidth is.

atdavie · Mar 29, 2007

internal speaker

Does anyone have it working over the internal speaker ?

If so why am I stuck with the sound coming through the external speaker ? Very annoying - especially as everything else seems ok.

Saying that bad line quality with VOIP provider :-( even through PC using headphones.

stevehead · Mar 29, 2007

UK Dialplan edit for Landline Numbers

Marvellous stuff - I've been waiting for the MS VOIP to come to life - Sweeet!

It all works as described, the only issues I can see are:
Incompatible dialling with existing phonebook numbers
Audio out from stereo speakers, not earphone on phone (Universal)

Here's something you can do to make dialling UK Landline (11 digit) numbers work directly, and from your existing Phonebook entries.

Say you have a number you want to dial that is 01752 265111
Your GSM Phonebook entry says 01752 265111, but to dial using SIP you have to use 0044 1752 265111, which is a pain. This is because SIP wants the full international dialling code. What we have to do is tell the SIP Phone to replace the first '0' in our UK Number with '0044' to make it work.

OK... first, download and install 'Total Commander'. Use this to edit the file in Windows called ipdialplan.xml. It's set to read-only, but Total Commander can deal with this.

Find the section 
By default, it looks like this:

<rule pattern='1\s*-?\s*(\d{3})\s*(\d{3})\s*-?\s*(\d{4})(\s*[Xx]\s*\d+)?'
dial='sip:91\1\2\3@$host$'
display='1 (\1) \2-\3'
transfer='sip:1\1\2\3@$host$'

Make it look like this:

<rule pattern='0\s*-?\s*(\d{4})\s*(\d{6})(\s*[Xx]\s*\d+)?'
dial='sip:0044\1\2@$host$'
display='0 (\1) \2'
transfer='sip:0044\1\2@$host$'

Save and reboot the phone.
You will now be able to dial normally, and the SipPhone will translate the number for you in the background, and dial the way it needs to, with the full UK international code.

I am using a service called SipDiscount - it works for this sip provider.
You can set up a 1 min free test account without paying or subscribing just to test out your Sip Settings.

Please someone fix the Audio routing to the phone earpiece, then we have a fully integrated SipPhone for free - have you seen how much they cost to buy?

Steve Head

Jwright 2.01.06wwe
Radio 1.13
XDA Exec

raix · Mar 29, 2007

I've it running with the internal Speaker and my BT Headset.
But i didn't change anything. Just installed it and it worked.
Perhaps it's a device-specific Problem?
I'm using a Hermes with XDA Live 0.20

papamopps · Mar 29, 2007

raix said:
I've it running with the internal Speaker and my BT Headset.
But i didn't change anything. Just installed it and it worked.
Perhaps it's a device-specific Problem?
I'm using a Hermes with XDA Live 0.20

Mine is HTC wizard - and yours atdavie???

atdavie · Mar 29, 2007

Internal speaker

I have a Wizard with an O2 badge, running Farias WM6 and the WM6VOIP cab kindly found on this thread.

stevehead · Mar 29, 2007

Better Number Display - UK Landline Dialplan Edit

stevehead said:
Make it look like this:

<rule pattern='0\s*-?\s*(\d{4})\s*(\d{6})(\s*[Xx]\s*\d+)?'
dial='sip:0044\1\2@$host$'
display='0 (\1) \2'
transfer='sip:0044\1\2@$host$'

Actually, change the display line to read:
display='(0\1) \2'

This doesn't effect the use of the phone, just the way the UK Number is displayed. It now looks like this: (01752) 256111

Sleuth255 · Mar 29, 2007

@eluth:

Where did you find your information on the VoIP configuration service provider? Could you post me a link to a definition plz?

@US users. How to mod your IP dialplan for 10 and 7 digit calling if your SIP provider isn't a PBX (ie you're using a Vonage SoftPhone)

The default 10 digit rule dials 9 first to get out then the 10 digit number. Sooo... under 10 digits rules change this:

Code:

        dial='sip:9\1\2\3@$host$'

to this

Code:

        dial='sip:\1\2\3@$host$'

To get the 7 digit rules working, you need to insert your own default area code (in the example, mine is 262)
Change these

Code:

        dial='sip:9425\1\2@$host$'
        display='\1-\2'
        transfer='sip:425\1\2@$host$'

to these

Code:

        dial='sip:262\1\2@$host$'
        display='\1-\2'
        transfer='sip:262\1\2@$host$'

eluth · Mar 29, 2007

Sleuth255 said:
@eluth:

Where did you find your information on the VoIP configuration service provider? Could you post me a link to a definition plz?

As I mentionned before, it's mostly standard OMA client provisioning, at the MS sauce. You get info in the Windows Mobile 6 SDK and the Windows CE 6.0 SDK.
The tricky parts:
- WM6 SDK is incomplete and does not describe the value string,
- WinCE 6.0 SDK has the wrong headers, some syntax issues, especially around &lt and &gt...

Based on that, some XML correction and trial/error did the rest.

--eluth.

mrmrmrmr · Mar 29, 2007

mrmrmrmr said:
is there any full documentation/samples on the provisioning xml file ?

I need to use STUN server with my SIP proxy. So I am looking for that parameter.
Also, I would like to change the codec to GSM if it's ever possible.
Any information regarding these parameters would help me much.

hi guys,

there are very nice findings on the dialplan.
However my stun and codec question ?
is it possible to use stun server ? is it possible to select codecs ?

muizmotani · Mar 30, 2007

eluth said:
Check if the VoIP is support is present in your ROM, as some cooks have removed them from the ROM.
so look for the Internet Calling today item as well as for the following files from the packages VoIP and VoIPOS in the \Windows directory:

[*]ipdialplan.xml
[*]dnsapi.dll
[*]voipphonecanvas.dll
[*]rtcdll.dll

...

For the HTC Wizard, the Orwell1984, the PDAViet ROMs, MB ROM and mUn_aRTM_10_2_0_8_WWE have the necessary modules.
For other devices, try it for yourself, as I don't know the specifics of the ROMs..

I have a wizard and I just installed the MBE 4MB ROM and I can't find the dll files nor do I have an Internet Calling today item. I don't want to download the cab file that someone else referenced in an earlier post that was used to install voip support because it seems to be a Hermes cab. Did I do something wrong or am I just missing something?

[GUIDE] How to configure VoIP/SIP client in WM6.

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