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.
I have seen numerous posts on this and alot of depressed people trying to get VOIP working with 3G on their trinity, after alot of forum reading and testing I have found a bulletproof solution. It lets you configure your SIP settings, you can make calls and receive calls and hear and people can hear you over wifi and 3G. However, I cannot get incoming calls to ring at all when using 3G only (it works with WIFI flawlessly). So basically unfortunately VOIP as it is now with wm6 only lets you make VOIP over 3G (you cannot receive calls as it just doesnt ring, I suspect it is a firewall issue as over wifi it doesnt ring till I open up ports on my router).
First install the first VOIP file, it has sip config built it. YAY lol
Now you need a cab extractor program like wince manager, which you can download for free [SIZE=-1]www.ocpsoftware.com/products.php?nm=cecabmgr
Now extract the rtc.dll file in the second cab I have attached and copy it to your windows folder and overwrite the one currently there. Without the dll file VOIP works but u cannot hear anything when the other party talks. Once you overwrite this dll, and u have to use Resco file explorer or similar, then VOIP works 100%. Ofcourse you have to run sipconfig, and enable VOIP over 3g and put in your SIP settings
Thats it. Pretty easy fix right
[/SIZE]
tks
Thanks my friend.
Nadavi
Thanks firstly
The second cab isnt containing any dll
Dear mcq3000
Can you share with us which VoIP provider you tested this method with, and can you post your dialplan?
Thanks
I did everything that is described in this thread.
On the today-screen it says my accountname and available. But how can I make a voipcall then?
I use MUNs 2.01 TE ROM.
ulischultz said:
I did everything that is described in this thread.
On the today-screen it says my accountname and available. But how can I make a voipcall then?
I use MUNs 2.01 TE ROM.
Click to expand...
Click to collapse
Just click once on the today plugin and "available" will change to "selected"
Then make a call as usual and you'll see your voip network name instead of the gsm carrier.
Thanks!
I reinstalled the files from above now, and now it only says on the today-screen:
"Internet Calling: No Service"
It seems, that voip does not connect at all now.
When I click on this line, it opens the Internet Calling settings page of the phone.
where is the interest of VoIP over 3G ? i think about cost ...
how many data is used for a conversation with VoIP over 3G ?
extenue said:
where is the interest of VoIP over 3G ? i think about cost ...
how many data is used for a conversation with VoIP over 3G ?
Click to expand...
Click to collapse
In Saudi Arabia, you can get unlimited HSDPA for 100 dollars per month.
Why I can't find any dll inside the wm5voip.cab ?
with skypho.net it doesn't work, it tells me "Skypho:no service", and if I make a phone call him it activates the gsm Does someone succeed us?
the rtc.dll was not cancelled but I has glued him above, this ok?
thanks for any suggestion
to open the cab and extract the file you need to use wince cab manager program. You cannot just open cab with winrar to see contents, those arent the real file contents. Get Wince cab manager and install it and open cab with that program and you will see dll and you can extract dll.
mcq3000 said:
to open the cab and extract the file you need to use wince cab manager program. You cannot just open cab with winrar to see contents, those arent the real file contents. Get Wince cab manager and install it and open cab with that program and you will see dll and you can extract dll.
Click to expand...
Click to collapse
in fact I have used wince cab manager,
can be that with Skypho.net it has in anymore this configuration stun?
Dominio voip.eutelia.it
Sip Proxy Server voip.eutelia.it
Sip Proxy Socket 5060
Stun Server stun.voip.eutelia.it
Stun Server Socket 3478
also tried with Stun but nothing.
and also activated the door 5060 from the register on 1 in voip sip.
Do I have to change telephone manager?
thanks still
alishankiti said:
In Saudi Arabia, you can get unlimited HSDPA for 100 dollars per month.
Click to expand...
Click to collapse
I've got unlimited HSDPA for 7 euros / month In the Netherlands
Floriz said:
I've got unlimited HSDPA for 7 euros / month In the Netherlands
Click to expand...
Click to collapse
Hotdiggididaaaaaaaam! That is pretty cheap!
mcq3000 said:
I have seen numerous posts on this and alot of depressed people trying to get VOIP working with 3G on their trinity, after alot of forum reading and testing I have found a bulletproof solution. It lets you configure your SIP settings, you can make calls and receive calls and hear and people can hear you over wifi and 3G. However, I cannot get incoming calls to ring at all when using 3G only (it works with WIFI flawlessly). So basically unfortunately VOIP as it is now with wm6 only lets you make VOIP over 3G (you cannot receive calls as it just doesnt ring, I suspect it is a firewall issue as over wifi it doesnt ring till I open up ports on my router).
First install the first VOIP file, it has sip config built it. YAY lol
Now you need a cab extractor program like wince manager, which you can download for free [SIZE=-1]www.ocpsoftware.com/products.php?nm=cecabmgr
Now extract the rtc.dll file in the second cab I have attached and copy it to your windows folder and overwrite the one currently there. Without the dll file VOIP works but u cannot hear anything when the other party talks. Once you overwrite this dll, and u have to use Resco file explorer or similar, then VOIP works 100%. Ofcourse you have to run sipconfig, and enable VOIP over 3g and put in your SIP settings
Thats it. Pretty easy fix right
[/SIZE]
Click to expand...
Click to collapse
Sorry - just looking for solid confirmation - been tearing my hair out to get this working. I install the VM VI Black VOIP & Config.CAB on my phone and replace the rtcdll.dll file once extracted from WM6VoIP.CAB using resco explorer. when you refer to fist and second files - it's better to specify file names to be sure we're all singing from the same songsheet!
mcq3000 said:
to open the cab and extract the file you need to use wince cab manager program. You cannot just open cab with winrar to see contents, those arent the real file contents. Get Wince cab manager and install it and open cab with that program and you will see dll and you can extract dll.
Click to expand...
Click to collapse
Thank you, I did use winrar
OK I did anything well, I can make a call, am heard speak but I cant hear what the recipent says. On other tries I hear him for 1 second. I also have too much noise on the speaker. While on the other side the recipient hear clearly
Could somebody post the rtc.dll from the WM6VoiP.cab???
I don't know how to extract this...
Thanks!
ulischultz said:
Could somebody post the rtc.dll from the WM6VoiP.cab???
I don't know how to extract this...
Thanks!
Click to expand...
Click to collapse
Here it is. As for me, this solution only worked for one call, then no cigar after that.
When i try to edit the registry to open the port it says
access is denied how to i get around it
Any help please.
A registry entry to open a port?
Mind to talk more about what are you trying to do actually?
If you are doing SIP, it's your application to open that port, not the registry.
I am trying to do sip but was told that the port had to be opened by
changing the registry. The item below is what i am trying to change
HKLM\System\VoIP\SIP
EnablePort5060 = 0
as i am unable to connect to a sip server.
Hiya Zed
Fancy meeting you here
So you ended up completing this by removing the memory card, rebooting (soft reboot) and making the change. It's certanly an option to remember to get the job done
Oh, Welcome to XDA Developers You can normally find me in the Hermes room
zed333 said:
I am trying to do sip but was told that the port had to be opened by
changing the registry. The item below is what i am trying to change
HKLM\System\VoIP\SIP
EnablePort5060 = 0
as i am unable to connect to a sip server.
Click to expand...
Click to collapse
This registry is only for RTC app which is suppesed to be in WM6 but is not. If you have instlled VoIP from installers taken from x-dev then OK, you can play with this registry, but if you are using some standard VoIP application like SJPhone or some other commercialy available one, then forget about this registry. Your app will do it for you.
However, to change registry, you need proper registry editor. like Resco in eg.
If yo have problem with VoIP and RTC then take in mind that there are SIP servers wth which RTC works, and some other SIP servers with wich RTC does not work. You can try with asterisk, FWD and some few more, but mostly with others you will have problems.
Easy solution, take SJphone for PPC, works well with all SIP servers, is free, but is bit to havy for the system. The best commercial VoIP app (at leas for me) is Aget phone, also good is X-PDA from GlobalIPTel.
Voip sip - working
http://forum.xda-developers.com/showthread.php?p=6348469
I am trying to get VoIP working on my wing, i've read around for days and can't seem to find anything specific to the wing, I have default rom, tried both SIP config tools both crash when I start them. I saw that there should be a voip dll in \Windows and there isn't, anyone have voip calling working with their wing?
requenym said:
I am trying to get VoIP working on my wing, i've read around for days and can't seem to find anything specific to the wing, I have default rom, tried both SIP config tools both crash when I start them. I saw that there should be a voip dll in \Windows and there isn't, anyone have voip calling working with their wing?
Click to expand...
Click to collapse
why allways starting a new thread????
http://forum.xda-developers.com/showthread.php?t=361002
As most of you have probably already figured out by now the Samsung Galaxy S II, despite running Android 2.3.3, doesn’t have the native Android SIP Stack integrated.
Normally, all smartphones running Android 2.3.3 or higher come with a native SIP stack, allowing handset owners to register their SIP accounts to make and receive calls through VoIP.
Unfortunately this feature has been deliberately dropped by the Samsung cowards who obviously bowed to the pressure of greedy telecom providers (we all know how adamant these sorts of companies are about having this feature disabled).
There are various VoIP clients available on the market (both free and paid) that you can use for making VoIP calls (Sipdroid, csipsimple, etc.) but I’d definitely recommend giving the original Android SIP stack a try as it works nicely and reliably.
Here are the full step by step instructions for enabling the native SIP stack on your Galaxy S II again (I think there’s already a similar thread out there by a member named jarsiv, but I decided to create a dedicated thread describing all steps thoroughly):
- Unzip the attached file “android.software.sip.voip.zip” and add it to “system\etc\permissions\". After you’re done, reboot your phone and voila you can enjoy the benefits of the native SIP stack again
By default, the native Android SIP stack only allows for VoIP calls placed via Wifi (it seems that even the mighty G from mountain view bowed to the telecom operators), but there is a way to enable support for VoIP calls via 3G as well:
- You need to modify your framework-res.apk to enable SIP over 3G by editing the xml file "res\values\bools.xml" and setting the parameter for "config_sip_wifi_only" to false.
In order to modify your framework-res.apk you’ll need to follow the following instructions (again this is only necessary if you want to enable 3G support and not required if you only want to place VoIP calls via Wifi):
- Download the program “APK Manager” (just search the forum) and install it on your PC
- Copy your original "framework-res.apk" (via adb push or root explorer) to your PC and place your original "framework-res.apk" into the "\place-apk-here-for-modding\" folder
- Launch "script.bat"
- Press 22 to set your project and choose "framework-res.apk" as your project by pressing 1
- After setting "framework-res.apk as your project, press 9 to Decompile the apk
- Edit the xml file "res\values\bools.xml" and change the parameter for "config_sip_wifi_only" from “true” to “false”
- Recompile the apk by pressing 11
- When asked whether it is a system file, answer “yes”
- When asked "would you like to copy over any additional files...?", answer “yes”
- The script will be temporarily paused and you shouldn’t press a key now.
- Now go to the “keep” folder (located in the same folder as "script.bat")
- In the “keep” folder delete the following file:
"resources.arsc"
- Go back to the "script.bat" screen and press a key to let the program finish the job
- Your new "framework-res.apk" has been created and can be found in the "\place-apk-here-for-modding\" folder (it’s called "unsignedframework-res.apk" and doesn’t work yet).
- After unzipping the entire content of the "unsignedframework-res.apk" you should have 5 objects (3 folders and 2 files).
- Now zip these files with Winzip choosing "No compression" as zipping method
- After zipping it, rename the zip into "framework-res.apk".
- Now you can use root explorer or adb push to replace the original framework.apk with your newly created "framework-res.apk".
- Reboot your phone.
Enjoy the native SIP stack with support for both Wifi and 3G
Note: If you’re sitting behind a NAT you might not be able to receive incoming calls after a while as the keep alive packets (even when set to “always send”) sent by your android phone don’t seem to suffice to keep the connection alive. Unfortunately the keep-alive interval can’t be changed manually in Android 2.3.3 meaning this issue has nothing to do with your Galaxy S II but rather with the Android native stack.
Good luck and enjoy your VoIP calls
Nice, but native SIP is using wrong mic.
So its pretty useless right now. I recommend CSipSimple (nightly ginger builds), which works excellent, is integrated as well and let you choose a better codec.
there's already a patch in work to fix the issue with the mic.
Btw I'm also a huge fan of CSipSimple and consider it to be the best open source SIP client out there, even surpassing sipdroid in terms of functionality and variety of settings.
yep, csipsimple (nightly builds) is WAY better than sipdroid or built-in sip!
i'm a heavy sip user and tried everything.... csipsimple is the best!
thanks. I am sure this is good stuff Dialling fully numeric numbers is possible if I set call settings to always ask whether to use internet calling, but the samsung dialer doesn't allow calling sip addresses in the internet call section of a contact.
Or am I missing something?
Here it is modified mi rom KE7
requires root root explorer or similar use, route to paste system / framework
perdon por mi ingles soy español
julianico said:
Here it is modified mi rom KE7
requires root root explorer or similar use, route to paste system / framework
perdon por mi ingles soy español
Click to expand...
Click to collapse
here is same modified from
[30.05.2011][CF-Root v3.5] KE2/7 - su+bb+CWM3
I can get outbound calling to work now. inbound doesn't work at all.
update: inbound does work, just when sgsii picks-up the dialling phone continues to ring as though the phone wasn't answered (though there is echo of audio picked up by the sgsii). Maybe my dialing phone has faulty setup? i dunno.
I tried to make this trick but I don't know if it gone well.
I changed my framework-res.apk for to usa SIP under 3G but in settings I see a menu called "Use Internet Call (only WiFi)" (this in translated from italian, I don't know what is the right menu in english) with three options:
- For all calls
- Only for calls via Internet
- Ask me
Are the changes of framework-res.apk gone wrong?
thanks. I am sure this is good stuff Dialling fully numeric numbers is possible if I set call settings to always ask whether to use internet calling, but the samsung dialer doesn't allow calling sip addresses in the internet call section of a contact.
Or am I missing something?
Click to expand...
Click to collapse
For local SIP to dial you need to select allow receive calls under settings. Drains your battery like crazy though.
m_pascal said:
I tried to make this trick but I don't know if it gone well.
I changed my framework-res.apk for to usa SIP under 3G but in settings I see a menu called "Use Internet Call (only WiFi)" (this in translated from italian, I don't know what is the right menu in english) with three options:
- For all calls
- Only for calls via Internet
- Ask me
Are the changes of framework-res.apk gone wrong?
Click to expand...
Click to collapse
It seems that you encountered some kind of error while editing the framework-res.apk because obviously you're stuck on the old settings. When I first attempted to add 3G VoIP calling I couldn't set it up successfully either but when I gave it another try I could eventually get it to work.
After successfully modifying your framework-res.apk you should see the following Internet calling settings:
Option "Use Internet calling":
- For all calls when mobile network available
- Only for Internet calls
- Ask for every call
You can also use one of the frameworks uploaded by julianico or crypticc and then it should work.
@crypticc are you located behind a NAT? What are your settings? What SIP provider are you using?
There are tons of factors that could potentially cause this problem but the more info you provide the more likely we can help.
mobilegalaxy said:
As most of you have probably already figured out by now the Samsung Galaxy S II, despite running Android 2.3.3, doesn’t have the native Android SIP Stack integrated.
Normally, all smartphones running Android 2.3.3 or higher come with a native SIP stack, allowing handset owners to register their SIP accounts to make and receive calls through VoIP.
Unfortunately this feature has been deliberately dropped by the Samsung cowards who obviously bowed to the pressure of greedy telecom providers (we all know how adamant these sorts of companies are about having this feature disabled).
There are various VoIP clients available on the market (both free and paid) that you can use for making VoIP calls (Sipdroid, csipsimple, etc.) but I’d definitely recommend giving the original Android SIP stack a try as it works nicely and reliably.
Here are the full step by step instructions for enabling the native SIP stack on your Galaxy S II again (I think there’s already a similar thread out there by a member named jarsiv, but I decided to create a dedicated thread describing all steps thoroughly):
- Unzip the attached file “android.software.sip.voip.zip” and add it to “system\etc\permissions\". After you’re done, reboot your phone and voila you can enjoy the benefits of the native SIP stack again
By default, the native Android SIP stack only allows for VoIP calls placed via Wifi (it seems that even the mighty G from mountain view bowed to the telecom operators), but there is a way to enable support for VoIP calls via 3G as well:
- You need to modify your framework-res.apk to enable SIP over 3G by editing the xml file "res\values\bools.xml" and setting the parameter for "config_sip_wifi_only" to false.
In order to modify your framework-res.apk you’ll need to follow the following instructions (again this is only necessary if you want to enable 3G support and not required if you only want to place VoIP calls via Wifi):
- Download the program “APK Manager” (just search the forum) and install it on your PC
- Copy your original "framework-res.apk" (via adb push or root explorer) to your PC and place your original "framework-res.apk" into the "\place-apk-here-for-modding\" folder
- Launch "script.bat"
- Press 22 to set your project and choose "framework-res.apk" as your project by pressing 1
- After setting "framework-res.apk as your project, press 9 to Decompile the apk
- Edit the xml file "res\values\bools.xml" and change the parameter for "config_sip_wifi_only" from “true” to “false”
- Recompile the apk by pressing 11
- When asked whether it is a system file, answer “yes”
- When asked "would you like to copy over any additional files...?", answer “yes”
- The script will be temporarily paused and you shouldn’t press a key now.
- Now go to the “keep” folder (located in the same folder as "script.bat")
- In the “keep” folder delete the following file:
"resources.arsc"
- Go back to the "script.bat" screen and press a key to let the program finish the job
- Your new "framework-res.apk" has been created and can be found in the "\place-apk-here-for-modding\" folder (it’s called "unsignedframework-res.apk" and doesn’t work yet).
- After unzipping the entire content of the "unsignedframework-res.apk" you should have 5 objects (3 folders and 2 files).
- Now zip these files with Winzip choosing "No compression" as zipping method
- After zipping it, rename the zip into "framework-res.apk".
- Now you can use root explorer or adb push to replace the original framework.apk with your newly created "framework-res.apk".
- Reboot your phone.
Enjoy the native SIP stack with support for both Wifi and 3G
Note: If you’re sitting behind a NAT you might not be able to receive incoming calls after a while as the keep alive packets (even when set to “always send”) sent by your android phone don’t seem to suffice to keep the connection alive. Unfortunately the keep-alive interval can’t be changed manually in Android 2.3.3 meaning this issue has nothing to do with your Galaxy S II but rather with the Android native stack.
Good luck and enjoy your VoIP calls
Click to expand...
Click to collapse
Very nice info. Perhaps we can change the keepalive in the smali sources? Just reading this made me think of it.
pulser_g2 said:
Very nice info. Perhaps we can change the keepalive in the smali sources? Just reading this made me think of it.
Click to expand...
Click to collapse
Pulser, that's an excellent idea and maybe we can make some other changes in the smali sources as well to offer end users even more options when fine-tuning their SIP settings?
As you may have noticed, the Android native sip stack doesn't allow users to choose their desired Codec, let alone prioritize them. I think the end users should at least be able to choose between the following codes:
G.711 (A-law and μ-law) (64 Kbps for good connections (e.g. Wifi))
G.726 (32 kbps)
G.729 (8 kbps for 3 G)
Despite supporting VoIP calls for handsets located behind a NAT (or otherwise we wouldn't have the keep-alive checkbox) the android SIP settings don't allow users to specify NAT related settings such as:
- Enter Stun server
- ICE
- DNS SRV
Settings for QoS and Echo cancellation are not present either, though I haven't experienced and echo problems during my test-calls yet.
I think these are the most important settings and if we could add them people will probably stick to the native client instead of downloading alternatives from the market that often cause problems with the native dialer of the phone
mobilegalaxy said:
It seems that you encountered some kind of error while editing the framework-res.apk because obviously you're stuck on the old settings. When I first attempted to add 3G VoIP calling I couldn't set it up successfully either but when I gave it another try I could eventually get it to work.
After successfully modifying your framework-res.apk you should see the following Internet calling settings:
Option "Use Internet calling":
- For all calls when mobile network available
- Only for Internet calls
- Ask for every call
You can also use one of the frameworks uploaded by julianico or crypticc and then it should work.
@crypticc are you located behind a NAT? What are your settings? What SIP provider are you using?
There are tons of factors that could potentially cause this problem but the more info you provide the more likely we can help.
Click to expand...
Click to collapse
Hi - thanks for replying.
My sip is pbxes.org. I also setup a test sip on another network to test the call to my PC. I have finally succeeded in getting incoming working by setting pbxes.org to direct via a ekiga.net trunk and setting incoming and outbound.
However, I don't believe that or the NAT is the issue - my question is more about how to physically make the call.
What I mean is that in my dialer I literally cannot enter the sip address to dial. Even if I setup the sip: detail in the internet call placeholder in my phonebook contacts there's no call button.
sip:[email protected] for example.
I have had some success with copy-and-paste into the dialer app, and maybe the NAT thing matters a little because it is more resilient over 3g than NAT - but copy and paste is not ideal.
crypticc said:
Hi - thanks for replying.
My sip is pbxes.org. I also setup a test sip on another network to test the call to my PC. I have finally succeeded in getting incoming working by setting pbxes.org to direct via a ekiga.net trunk and setting incoming and outbound.
However, I don't believe that or the NAT is the issue - my question is more about how to physically make the call.
What I mean is that in my dialer I literally cannot enter the sip address to dial. Even if I setup the sip: detail in the internet call placeholder in my phonebook contacts there's no call button.
sip:[email protected] for example.
I have had some success with copy-and-paste into the dialer app, and maybe the NAT thing matters a little because it is more resilient over 3g than NAT - but copy and paste is not ideal.
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You’re right, NAT is definitely not the issue here as the NAT settings don’t affect the android dialer at all. I can’t enter a sip address in my dialer either and it appears to me that this feature has been intentionally disabled to deter people from calling a SIP target. I’m looking to find a remedy for this issue though.
In the meantime you can assign a SIP uri to a contact and then copy and paste it into the dialer and then you should be able to call the desired SIP uri. Alternatively you can also download a third-party app from the market (e.g. acrobits) which allows you to enter SIP uris directly without having to rely on copy and paste.
mobilegalaxy said:
You’re right, NAT is definitely not the issue here as the NAT settings don’t affect the android dialer at all. I can’t enter a sip address in my dialer either and it appears to me that this feature has been intentionally disabled to deter people from calling a SIP target. I’m looking to find a remedy for this issue though.
In the meantime you can assign a SIP uri to a contact and then copy and paste it into the dialer and then you should be able to call the desired SIP uri. Alternatively you can also download a third-party app from the market (e.g. acrobits) which allows you to enter SIP uris directly without having to rely on copy and paste.
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Click to collapse
good call, will check it out. Thanks
Sent from my GT-I9100 using Tapatalk
You’re right, NAT is definitely not the issue here as the NAT settings don’t affect the android dialer at all. I can’t enter a sip address in my dialer either and it appears to me that this feature has been intentionally disabled to deter people from calling a SIP target. I’m looking to find a remedy for this issue though.
In the meantime you can assign a SIP uri to a contact and then copy and paste it into the dialer and then you should be able to call the desired SIP uri. Alternatively you can also download a third-party app from the market (e.g. acrobits) which allows you to enter SIP uris directly without having to rely on copy and paste.
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good call, will check it out. Thanks
Sent from my GT-I9100 using Tapatalk
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Just a thought. On non samsung phone, long press menu brings up keyboard no matter what you are doing, if this could be enabled then it wouldn't matter what the phone keyboard allowed
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Hey there,
for everyone who want to use the native android SIP stack,
i made an update.zip for CWM.
Important:
This Version is for WiFi only!
To activate using SIP via 3G, please follow first Post, after flashing my Native_Android_SIP.zip.
Just download the attach. and flash it with CWM.
Done!
crypticc said:
Just a thought. On non samsung phone, long press menu brings up keyboard no matter what you are doing, if this could be enabled then it wouldn't matter what the phone keyboard allowed
Sent from my GT-I9100 using Tapatalk
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I had the same thought and some VoIP clients actually support this feature. As far as I know some roms enable this feature too since many non-Samsung phones bring up a keyboard if you long press or double click the input field in the dialer app...
Also there are some dialers on the android market which might support this feature. I'm going to give some of them a try and let you know.
@BMWpokerAce thanks for the file
KE8
crypticc said:
here is same modified from
[30.05.2011][CF-Root v3.5] KE2/7 - su+bb+CWM3
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Hi,
Great work and thanks to the OP for this.
Will the modified one for CFroot KE7 work with CFRoot KE8?
( I have Lightning 1.2 installed).
I tried to enabled the 3g/sip myself, but I don't have Windows but Linux instead.
I did attempt this myself by unzipping the framework-res.zip but the bools.xml is nowhere to be found, along with the values directory.
Any ideas for modding this on Linux? (I know, someone will suggest using WINE ).
Code:
[email protected]:~framework-res.extract/res# ls
anim drawable-en-hdpi drawable-nodpi layout-port raw-da raw-es raw-hu raw-nl raw-tr xml-land
color drawable-hdpi layout raw raw-de raw-fi raw-it raw-pl xml
drawable drawable-land-hdpi layout-land raw-cs raw-en-GB raw-fr raw-ko raw-ru xml-en
[email protected]:~framework-res.extract/res# cd values
-su: cd: values: No such file or directory
This gives nothing:
find -name bools.xml
PS. Although I see there is a point for SIP, I cannot understand why I would use it. From what I see, one has to create a SIP account somewhere with some company and pay for this. SIP is a standard, but from what I understand I have to connect with a third party server ($$s); I imagine someone has to pay for the phone number e.g There is OnSip.com, but the prices are very steep for the average residential client. Businesses can afford this, but my monthly mobile phone bill inclusive of data SMS and calls is about $20, but OnSip's cheapest is $40 yet only for a SIP account.
Or did I miss the point? Please would someone persuade me its a good idea? ( I already have Skype)
forgetmyname said:
H
PS. Although I see there is a point for SIP, I cannot understand why I would use it. From what I see, one has to create a SIP account somewhere with some company and pay for this. SIP is a standard, but from what I understand I have to connect with a third party server ($$s); I imagine someone has to pay for the phone number e.g There is OnSip.com, but the prices are very steep for the average residential client. Businesses can afford this, but my monthly mobile phone bill inclusive of data SMS and calls is about $20, but OnSip's cheapest is $40 yet only for a SIP account.
Or did I miss the point? Please would someone persuade me its a good idea? ( I already have Skype)
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Click to collapse
The point of SIP clients is for business users primarily, for example, I can use it to login to my SIP phone when I'm walking around the office and if someone phones me, my mobile will ring, if I want to make an internal call, I simply dial an extension.
For a personal user, it's pretty much pointless unless you're running a SIP server at home.