Now that there is a solution for VoIP available I can't help but get greedy So far I used a softphone with multi provider support for this task, but now I wonder: Is it possible to use two different providers at the same time? Right now I am helping myself by switching by hotkey assigned cab files (which works pretty well btw), but what I am really looking for is using one provider for inbound calls and the others for outbound telephony. There are of course providers which can act as a proxy for others, but that solution never really worked out for me as many voip vendors seem to block these attempts (especially pointing a finger at SipGate here ). So, is there any solution for handling multiple providers?
Theoreticall, they are "Backup SIP Settings" you can provision in Windows Mobile 6. Just look at http://msdn2.microsoft.com/en-us/library/aa926605.aspx.
Having only one working VOIP provider, I haven't tried this yet.
However, the format of the XML provisioning is very similar to the one I posed in http://cleanimport.xda/index.php?posts/299950/.
You just have to replace parm name="SIPSettings" by parm name="BackupSIPSettings".
If you try it out, please report here.
Thanks.
--eluth.
The CAB installs fine, but there is no difference between installing a new provider with just "SIPSettings" and "BackupSIPSettings" I tried this with Arcor and SIPGate to test ut whether both incoming numbers would be available at the same time.
Related
Hi
I am new to this form.
I had a look at the tsril.zip example as it seems from this forum that to make/receive a data call the ril is the only option. Unfortunately the documentation for ril is nonexistent and I want to ask if someone has done an example of a client/server data call, data transfer between to XDA's.
Thanks in advance.
Anton
It all depends on what type of data you want to transfer. I have never used RIL so I don't know the advantages of it, over the normal programming interface. I say normal, not thinking there is anything wrong with RIL but the little I do know is that RIL is operating system level and is normally hidden from the application programmer. There must be times when RIL must be used if Microsoft forgot functions in the normal APIs.
However I would say that for a data link between 2 devices then RAS seems the way forward. One device would be the Server and the other the Client. I have only ever programmed RAS from the client side, making a Telnet link to an already running server.
I just had a look at the help on Embedded Visual C++ 3.0 and under 'Creating a TCP Stream Socket Application' it explains the server side and then links to the client side. Have a read there.
RAS is not an option as it is a server consept, which I think does not even exist as a service on the desktop windows os let alone on a Pocket PC.
What I am looking for is something similar that you would be able to do with a landline modem or a gsm modem connected to a serial port of a pc (even a usb modem) which is to send AT commands to make/accept a call and then to make use of the serial port as if it is an io stream. This is possible on windows, linux using either c#, c++ or java. I can even do this with an embedded gsm module like the siemens TC45 java module.
What I can not understand is why Microsoft and the XDA suppliers (I am using a Qtex running Pocket PC 2002) is making it so difficult to make/answer a data call and let you run your own protocol over the connected stream.
RAS (Remote Access Services) is built into all Windows Operating Systems, including Windows CE. Did you look up the help example I pointed to ? What you do with RAS on an application level is for you to code but the connection itself is handled by the operating system.
As to making a connection similer to a modem and using AT commands. Then no you can't use AT commands directly, but you don't need to. To get that type of connection you need to use TAPI. Once Tapi has made a DATA_MODEM type connection, you use the TAPI callback connect event to then ask for a file handle that you can use with the normal WriteFile and ReadFile commands.
I already have something on this forum about that see:-
http://forum.xda-developers.com/viewtopic.php?p=7857&highlight=#7857
The problem is not so much making the data call but it is accepting/answering the data call that I can not get working on the XDA
Not sure if this is your problem, but in Australia they have a seperate "data" phone number for the same SIM. If yo call one number you get voice, if you call another number you get data on the same sim. Not sure if this may be effecting you.
I just looked into all this ras stuff a bit deeper, and yes I think I may of been talking rubbish. Although RAS does exist in PPC2002, it can't see any functions that allow the device to answer the line. I can see that PPC2003 has a Ras Server and RasIOControl that looks like the answer there.
Also looking into Tapi, I can't see how you answer a data call using that.
So I now understand your problem. It seems PPC is made as a client device.
Is GPRS the answer ? With that the link is always present and then you can use the built in Ras functions on seperate devices. Since you only pay for data actually sent then would it matter if the network link was always present ?
I have only worked out Tapi and Ras from the client side so I have never had this problem, but I agree it is an interesting one. I will have a bow out of this thread for a bit and see if anybody else knows how to Answer a Call.
Thanks for trying.
GPRS is also not the answer as you need a server in between that both XDA's can connect too and use as a router as the XDA's do not get fixed IP adresses but actually a NAT adres from the APN so you can not connect directly between the two.
Now that's an interesting thought, how about using some free web space to act as a pigion hole for your data ?
Believer: A seperate number? How do you know this, and what would the number be?
I can actually make two XDA connecting to each other using TCP. The trick is that I have one client always connect to a server to register it IP address with an ID. Then the caller send a query to the server to look up the callee's IP address.
In this way, accept() and connect() work fine.
I am using AT&T network and not sure about if other networks behave the same.
I wrote two application using TAPI. One is ModemDial and the other is ModemWatch, if I make a voice call the ModemDial dials the number successfully and ModemWatch reports that an incoming call exist. But when I change the behavior of call to DATAMODEM, the ModemWatch couldn't track any incoming call.
Is there any one, had some experience with pick a call for DataModem?
Best regards,
A. Riazi
riazi said:
I wrote two application using TAPI. One is ModemDial and the other is ModemWatch, if I make a voice call the ModemDial dials the number successfully and ModemWatch reports that an incoming call exist. But when I change the behavior of call to DATAMODEM, the ModemWatch couldn't track any incoming call.
Is there any one, had some experience with pick a call for DataModem?
Best regards,
A. Riazi
Click to expand...
Click to collapse
In your ModemWatch application make sure you are calling lineOpen with dwPrivileges (the 7th parameter, 1 based) equal to LINECALLPRIVILEGE_OWNER and dwMediaModes (the 8th parameter) equal to LINEMEDIAMODE_DATAMODEM.
If you're able to answer a voice call successfully, making these changes to the ModemWatch application should allow you to answer a data call.
May I know the Buildin VoIP of WM 6.0 support the service type of Nortel VoIP gateway? I using HKBN now, but it only show Searching...
VoIP is still very much developer stuff. We can't give you the "works with VoIP" list because we are still sorting why different users are experiencing different results with the same VoIP provider. Your best bet is to read and understand the "{Solved} How to configure VoIP/SIP client in WM6" thread. From that point, you will need to find the exact settings for your Nortel switch. Its service must be SIP based and have DNS referenced Public IP addressing.
Hello everyone!
Situation: I have several VoIP provider (sipgate, 1und1...) My problem is I can only setup a single provider in my phone with built-in WM6 VoIP client.
Solution attempt: I set up an account with pbxes.org. There I added my SIP providers and an extension with username and password. Now I tried to set this up using the Sip Config Tool V2.0.1. Here are the settings:
Sip Server: pbxes.org Port: 5060
(tried others like sip.pbxes.org(com) and the IP)
username: username-extension_number
password: password
That's it. It's not working. The Today Internet-Calling-plugin tells me "searching....", but never to "selected"
I also tried using x-lite on my desktop pc. With the same settings I successfully registered and I was able to use pbxes just as it is supposed to be.
When using sipgate directly with WM6 it works, but not with pbxes.
I am kind of running out of ideas. So any help is really appreciated.
Thanks in advance,
Stefan
Tried several times as well on my S620.
Now, two days ago got a S710 and was curious to test VOIP on it.
Very first time I tried I was thinking everything was fine, at least nearly.
The Today plugin showed "Searching" (like with providers it works with)
The icon for Internet Phone showed up.
Unfortunately it "semi worked" only that time, any other tests I tried the Today Plugin didn't even get into "Searching".
The bad thing is that it did work once, like meaning it could work. Now PBXES status shows as Registered but it isn't actually. The other bad thing is that I tried to post in their forum for help but didn't receive much info back.
Bad luck, but I'll try and try again. Maybe newer files one of these days and our probs would be solved.
Ka.
Subscribe message
I ever check the message from sip server, I found that windows mobile voip client not only send register message, but also try to send subscribe message, I guess that it is reason that some sip server only may work as registra, and not handle correctly subscribe. it seems the reason the client always display "searching", I will try to modify sip server to support subscribe message in the future.
try fring
try using fring...
it supports multiple sip accounts.. and actually register u not just subscribe you.
You are right, Fring or any other 3rd party app connecting to a SIP server can work maybe better than the "builtin" SIP functionality but personally would prefer being able to use a solution where I dial like usual and calls get automatically router to VOIP, not having to switch to a separate app.
This is why I'd rather like the builtin sip stack to work on its own with any SIP provider. I tested it with PBXES but doesn't work. I'm currently renting a VPS with Asterisk (Trixbox) installed and it does connect. Only prob is that even if it manages to register, seems MS VOIP implementation is still not really compatible or suitable for the use I wish to have.
Ka.
This is an old thread but this is the same problem I have on my HTC HD2 (Leo).
Installed sip drivers and voip config cabs and configured it with my account through pbxes and it is not working.
Can someone help?
Android (Moto droid, HTC Hero) I had no problem with pbxes and sipdroid...
Hi Guys,
Looking for some VOIP/SIP clients for HD2, anything other than PortSIP, Fring or AGEMobile would be preferable!
Going overseas tomorrow and my PortSIP stopped working and wont start working again so trying to find a program that's just as good AND easy to setup as PortSIP (Agemobile confused the hell out of me)
thanks,
Zhuk86
+1 I am in a similar boat. Looking for a stable VoIP solution to work on my HD2.
Ideally, I would love tight integration with the standard WinMo dialer.
+1. Same here!
sip client for hd2
need also a stable sip sofrware for hd2 +1
thnks for help
I tried PortSIP recently on my phoneset, and it works fine. However, as in my grief with all these SIP clients products, I fancy very tight SIP integration with existing Phone Dialer, with the option of editing Dial Plans to define outgoing call rules (was it IPDialPlan.xml ?). I heard this is how it used to be with WM5, but somehow operators have scrapped it out of ROMs in WM6.5.
Ideally, I should just dial my telephone numbers from phone, and based on the dial plan, the phone should direct calls either on SIP or Mobile Network. Am I asking for too much here?
[Loud Thinking]
I am going to bring my trusty old O2 XDA Exec out and put WM5 back on it, and then see if the SIP client can be (easily) configured on it.
bym007 said:
I tried PortSIP recently on my phoneset, and it works fine. However, as in my grief with all these SIP clients products, I fancy very tight SIP integration with existing Phone Dialer, with the option of editing Dial Plans to define outgoing call rules (was it IPDialPlan.xml ?). I heard this is how it used to be with WM5, but somehow operators have scrapped it out of ROMs in WM6.5.
Ideally, I should just dial my telephone numbers from phone, and based on the dial plan, the phone should direct calls either on SIP or Mobile Network. Am I asking for too much here?
Click to expand...
Click to collapse
+1 for me too!
Check my signature for an alternative voip client
I'd like to save everyone the trouble of figuring out how to get this done by sharing my setup with you. The VOIP itself works great over 3G/4G and Wifi, but to use the VPN you'll need to be on Wifi, 3g VPN doesn't seem to work.
Requirements:
SipDroid app in the market
A free account at pbxes.org
A paid account at callcentric.com (~$.019 a minute to call anywhere in the US)
A paid account at SuperVPN.net ($4 /mo if you pay for the whole year)
First set up a pbxes.org account, and connect to it with the SipDroid app, I recommend using this guide to walk you through the process.
http://guardianproject.info/2010/05...e-mobile-phone-system-for-android-and-beyond/
Once you have that working there is one crucial adjustment to be made within SipDroid. For some reason it comes default with all sorts of audio codecs, but only ONE of them seemed to work on the EVO, the Speex codec. So go into audio codecs and switch everything but speex to "never".
After that you should have a working VOIP system but you'll still need some kind of trunk if you want to make outgoing calls to land lines or cell phones. There are many solutions for this but I recommend callcentric.com, they seem to be the most recommended for this type of setup, and they worked great for me. You can pay $20 a month for unlimited US calling, this means you can be anywhere in the world and call the US for just $20 a month. Or you can prepay (this is what I did), then you pay a flat rate of about $.019 a minute to call the US from anywhere, and if you reach you pre paid limit, it just cuts off until you recharge it.
Once you have your callcentric account purchased, just go into your pbxes.org admin area and under trunks add one for call centric, use your callcentric # as the username, and callcentric.com as the sip server. Then go under Outbound routing, add a new one, name it whatever and choose your callcentric trunk from the pulldown menu, submit the changes and you're done.
Now you should be able to successfully make outbound calls to anywhere using SipDroid.
Lastly, this was the most challenging for me, the VPN. Apparently android, including 2.2, has some major issues with maintaining vpn connections, especially when you try and use them for VOIP. There is a huge issue queue in the android google groups forum where the problem is openly discussed without a real solution. BUT, while it appears the majority of VPN connections will fail, they CAN work if you get it set up just right. Setting up VPN's, specifically VPN's tailored for mobile devices, is not something I know how to do. In the android group thread someone mentioned SuperVPN.net as a working solution, I checked it out and sure enough they work great, I had zero problems with them the whole time I was out of the country.
So create a supervpn.net account, and then on your phone go into Menu -> Wireless & Networks -> VPN -> Add VPN -> Add PPTP VPN, create the connection and you are good to go.
*I didn't set up an inbound call # with callcentric as I didn't need one, I assume after you upgrade your callcentric account, adding the inbound trunk is similar to the outbound. Be sure and look into getting a free inbound number from sipgate.com before you go and pay for one, you'll be locked to a California area code, but free is free.
An alternative I use is having an Asterisk server at home and use IAXAgent from the market. IAX does not have the problems that SIP does when going over NAT. I can make calls over 3G or wifi. A lot of SIP providers also provide IAX accounts. IAX is just a better way to go for making calls over the Internet. SIP is excellent for the LAN.
ChrisDos said:
An alternative I use is having an Asterisk server at home and use IAXAgent from the market. IAX does not have the problems that SIP does when going over NAT. I can make calls over 3G or wifi. A lot of SIP providers also provide IAX accounts. IAX is just a better way to go for making calls over the Internet. SIP is excellent for the LAN.
Click to expand...
Click to collapse
Oops.. the method I posted actually works on 3g and 4g also, it's only the VPN that requires Wifi, I had worded it incorrectly, now it's fixed. (thank you)
I looked into setting up an asterisk server, but I didn't want to have to depend on my own server or home connection being available whenever I needed it, especially when I was traveling for more than a week.
Is IAX the same as a trunk, does it cost anything to connect to land lines or cellphones?
True, you method does make SIP work because you are using a VPN. IAX is an alternative to SIP. It is NAT friendly, and as long as the port is not blocked, it just works. Though, there are a fewer choices for clients compared to SIP. IAX was created by the Asterisk team. I do not know of any VOIP systems that support IAX, bug that does not mean they dont exist. I am a heavy Astersk guy, so IAX was my cup of tea.
What advantages does this have over google voice?
I'm curious cause i'll be going to england soon and would be nice to make calls over wifi.
ShoxV said:
What advantages does this have over google voice?
I'm curious cause i'll be going to england soon and would be nice to make calls over wifi.
Click to expand...
Click to collapse
None, in fact, it his disadvantages (See below). Also, most businesses, schools, etc. will block just about every VPN method. OpenVPN is the most flexible one I have found, which might be able to sneak around by using alternate sub-1000 ports (which most places won't block, since they require root access on whatever server they're running from).
OP: Might wanna take a look at this...
http://www.mywot.com/en/scorecard/supervpn.net
drmacinyasha said:
None, in fact, it his disadvantages (See below). Also, most businesses, schools, etc. will block just about every VPN method. OpenVPN is the most flexible one I have found, which might be able to sneak around by using alternate sub-1000 ports (which most places won't block, since they require root access on whatever server they're running from).
OP: Might wanna take a look at this...
http://www.mywot.com/en/scorecard/supervpn.net
Click to expand...
Click to collapse
Not sure what you're talking about, it saved me hundreds of dollars in roaming minutes while I was in Belize. Some places do block vpns I'm sure but I never had any issues, but you don't need the vpn itself unless you're in a country that blocks voip altogether, at which point occasional vpn is greater than no vpn.
Also supervpn was the only method I found that actually works on android, I think the risks referenced in that link you posted really only apply to desktop vpn use, not phones. Openvpn is great for somewhat advanced users and if you have a computer you can depend on as a server while you're out of the country for days or weeks, this guide isn't meant for someone capable of managing that.
As for Google voice, it just initiates an inbound call to your actual cell number, which does zero good when you're trying to avoid roaming. Now the new gmail implementation of voice shows promise as an actual voip solution, but currently that version is desktop only from what I can tell.