[Q]Native SIP Client on SGH-T989 - T-Mobile Samsung Galaxy S II SGH-T989

Hello Everyone.
This is my first post and I would like to say that I am loving this phone.
I used to have a Motorola Milestone running CM7 and it had built-in a Native SIP client under Call Settings.
Does anyone know how to modify the stock T-Mobile ROM to have this mod?
I tried the topic below already but I had no luck. Recompiled framework-res.apk as it is described but I still can't see Internet call options anywhere:
http://forum.xda-developers.com/showthread.php?t=1109962
If someone has the experience on doing this I would really appreciate if you could guide us on how to do this on our Phone.
Thanks!

Native SIP Client on SGH-T989
For everyone,
I noticed after performing the steps in the link I posted above, I do not see Internet Calling Settings under Call Settings, however, if I setup an Internet call address in my contacts and try to call it, the phone prompts me a popup asking if I want to add a SIP account...
So the SIP stack is really there but I just can't access it.
This may need some mod over Call Settings Menu to include Internet Calling Settings under it.
As soon as I figure out more stuffs I will post here.
If anyone knows a bit about it, please post here as well.
Thanks!

Thanks for the update on this, I actually switched to CM7 specifically because I couldn't find this functionality in other ROMS for this phone (coincidentally I came from a Milestone as well). If you can't find the native stack but are otherwise happy with your ROM, check out SIPDroid in the Market. It's a little less integrated, but works well and has some neat features of its own.

Other SIP Clients
germania said:
Thanks for the update on this, I actually switched to CM7 specifically because I couldn't find this functionality in other ROMS for this phone (coincidentally I came from a Milestone as well). If you can't find the native stack but are otherwise happy with your ROM, check out SIPDroid in the Market. It's a little less integrated, but works well and has some neat features of its own.
Click to expand...
Click to collapse
Well, So far I've tested CSipSimple and SipDroid as well... SipDroid is ok but I think the sound quality is not as good as the Native SIP Client.
CSipSimple has lots of bugs with our ROM... For example if you receive a call and try to answer it by sliding from left to right doesn't work.
It would be really nice if T-Mobile + Samsung wouldn't have cut this off from our firmware...

Related

Voip package available?

I'm not sure but I read through these threads:
http://forum.xda-developers.com/showthread.php?t=299950&highlight=voip
http://forum.xda-developers.com/showthread.php?t=303493&highlight=voip
Is there a seperate WM6 VOIP cab available somewhere? I don't see the suggested files, so I assume that I don't have it on my WWE WM6 rom.
Try search for VOIP...or look through todays postings for the same question:
and try these two posts
http://forum.xda-developers.com/show...highlight=voip
http://forum.xda-developers.com/show...highlight=voip
A quick search for VOIP will give you same answers
epmulligan said:
Try search for VOIP...or look through todays postings for the same question:
and try these two posts
http://forum.xda-developers.com/show...highlight=voip
http://forum.xda-developers.com/show...highlight=voip
A quick search for VOIP will give you same answers
Click to expand...
Click to collapse
Well, as you may have noticed I've found those threads, but I saw that the most of the thread is regarding HERMES.
I was wondering if anyone with an Artemis has been able to turn on VOIP (native voip)
I did manage to install the package as mentioned in the threads above, as well as put in the right config files. Also, the "Internet Calling: Available" today plugin is correctly displayed. However, I cannot switch dial pads and I don't get the correct status (connected).
Same problem here
Setup was done with Sip Config Tool 2.0.1
The plugin is configured : "Always, if available"
When connected by wlan or activesync the plugin displays : "Kein Empfang"
Disconnected from network displays : "Available"
Hopefully someone can help me
Guys I've got it working just fine on an xda orbit. There does seem to be a little bug though and that is when it's says "Searching" and "Available" on the today screen. Available actually means searching and searching means available
When searching is displayed, click on it in the today screen and you might notice a slight change in the signal which means it has switched to voip rather than gsm. Go to the phone screen and you'll see that the network has changed to your voip provider and any call you make will be routed through them.
I have downloaded sjphone yesterday and configured it to use my sip voip provider and it works brilliantly. The sound quality in sjphone is absolutely brilliant compared to the native voip, you should try it.
Finally it's working...
I tried to deinstall the plugin and after prompting an error it worked
The homescreen plug signals "searching" but as you said before it means "availabe". There are still a couple of things:
1) no incomming calls... i can't find the stun settings
2) +4930xxxxxx format doesn't work, i've to type 030xxxx
3) doesn't work after standby
Which version of sjphone are you using? There seems to be nothing available for wm 6
Hi BlueD,
I'm glad it's working for you. I tested incoming calls and that worked nicely as well without configuring any stun. The voip I used were sipgate and callcentric.
There's a post somewhere on one of the big voip threads and the guys managed to customize the dialing plans so that you can dial without using the international format. Do a quick search on voip dialing plan and you should find it somewhere.
Well obviously it doesn't cause the wifi disconnects, hence making it only useful for dialing out rather than receiving calls, except ofcourse if you want to leave the device on all the time, but that's not a very feasible solution.
The sjphone I downloaded was the one for v.1.60.303c Pocket PC 2003SE, it works nicely with excellent call quality. I also think you don't really need native voip capabilities to run that, which makes me think we could have had this on WM5 hmmm...
OK.. just fyi - i got a little bit more success in using the native VOIP solution. I had to flash from the WWE 6 to the BnB 2 ROM, then install the VOIP.cab and SipConfigTool. Now i can receive calls, but cannot place any... not sure why. Also, when i receive a call the sound is CRAP.. probably some codec issue here.
What type of internet connection have you?
Have you open ports on your router necessary for SIP, inbound and out?
epmulligan said:
What type of internet connection have you?
Have you open ports on your router necessary for SIP, inbound and out?
Click to expand...
Click to collapse
Yep I did. I got this confirmed by testing it with a softphone on my PC, which works great (same voip provider etc).

Windows Mobile 6.1 or 6.5 Professional ROM With SIP Enable

Hi all , I'm looking for WM6.1or 6.5 pro with VOIP/ sip client enable. Can someone please point me in the right direction to find this ROM or how to make my own ROM using another phone ROM that already have the features I’m looking for.
jcayse said:
Hi all , I'm looking for WM6.1or 6.5 pro with VOIP/ sip client enable. Can someone please point me in the right direction to find this ROM or how to make my own ROM using another phone ROM that already have the features I’m looking for.
Click to expand...
Click to collapse
Almost NONE of the Excalibur ROMS have VoIP cooked in. In order to use a SIP, you need VoIP actually cooked in. It cannot be installed. If you want a custom cooked ROM with VoIP, let me know and we can figure something out...
ookba said:
Almost NONE of the Excalibur ROMS have VoIP cooked in. In order to use a SIP, you need VoIP actually cooked in. It cannot be installed. If you want a custom cooked ROM with VoIP, let me know and we can figure something out...
Click to expand...
Click to collapse
Sure if you don't mind helping me that would be great. I'm just looking to use VOIP on my excalibur using the Wifi and data package. It would be nice to learn how to cook a ROM with that feature. I'm shocked no one has done one yet.
SIP on dash Works NOW!
I have Dash with with Ricky's 6.1 Rose 1.2.
After searching forum for other phones I've found one CAB file that adds SIP functionality to the Dash. Apparently SIP stack is part of WM 6.1 and later. This thing just enables that. It integrates so nicely it even adds icon next to the antenna signal. Sounds nice and clear with very minor latency (comparing to other portable SIP devices). It appeared to support STUN, but I ran it only on local home net. Donkow if STUN actually works right.
I did tried it with other ROMs on DASH and it allays worked.
It's not perfect though: Main problem is audio out settings: as with other applications output newer to the internal headphone but to the speaker or external one. I've found no way to overcome that. Another quirk is creation of dial plan - not straight forward. You also have to create account settings file externally and upload it each time. (not biggie as it done only once). There are application to do it on the phone, but I wasn ot able to run it successfully on Dash.
I disable all dial plan options and pass everything through, letting my Asterisk do the job. I also disable all power management and leaving WiFi on all the time. Surprisingly there is no major hit on battery life, it last whole day or two with SIP and WiFi on all the time.
I'm expecting that someday somebody will polish this option and include it as standard in every ROM. As a fact for me to be happy with SIP, only output must be fixed, everything else is tolerable. It will make DASH nice little SIP phone.
sipcipro said:
I have Dash with with Ricky's 6.1 Rose 1.2.
After searching forum for other phones I've found one CAB file that adds SIP functionality to the Dash. Apparently SIP stack is part of WM 6.1 and later. This thing just enables that. It integrates so nicely it even adds icon next to the antenna signal. Sounds nice and clear with very minor latency (comparing to other portable SIP devices). It appeared to support STUN, but I ran it only on local home net. Donkow if STUN actually works right.
I did tried it with other ROMs on DASH and it allays worked.
It's not perfect though: Main problem is audio out settings: as with other applications output newer to the internal headphone but to the speaker or external one. I've found no way to overcome that. Another quirk is creation of dial plan - not straight forward. You also have to create account settings file externally and upload it each time. (not biggie as it done only once). There are application to do it on the phone, but I wasn ot able to run it successfully on Dash.
I disable all dial plan options and pass everything through, letting my Asterisk do the job. I also disable all power management and leaving WiFi on all the time. Surprisingly there is no major hit on battery life, it last whole day or two with SIP and WiFi on all the time.
I'm expecting that someday somebody will polish this option and include it as standard in every ROM. As a fact for me to be happy with SIP, only output must be fixed, everything else is tolerable. It will make DASH nice little SIP phone.
Click to expand...
Click to collapse
Thanks for the info Sipcipro. Can you please post your intallation steps as well as the CAB file and or the CAB file link that adds SIP functionality to the Dash.
FYI
I successfully installed WM6VoIP.CAB which enables the Internet tab under the phone settings. However installing a sip config tool like, destr0_VOIP_WM61___SIP_Config.CAB , SipConfigTool_2_0_1.CAB or Setup_VoIP_v0.3.cab it appears that these sip config tools require WM professional
The file you have is the same one I have.
As I said before, configuration tools do not work for me, but it is possible to edit account config file offline, pack it and upload to the phone. Guide is here:
http://forum.xda-developers.com/showthread.php?t=299950
Good luck!
Let us know if you manage to resolve speaker redirection. This problem common for other applications like Skype and Fring. There got to be solution to that.
Cheers!
Ookba let me know when you would like to start working on cooking a Excalibur ROM with VoIP or if you found a cooked Excalibur ROM that have that feature alreadyand works good.

WM 6.5 SIP Client w/ Dialpad and Mute Function

Hi Guys,
I currently use free SIP service and google voice together with onedialer to make phone calls through Data. I currently use FRING as the SIP client, though it works good, but it doesn't have dialpad...so i can't enter tone responses (the windows dialer's dialpad tone sounds for some reason also doesn't work during SIP call) and neither does it have mute functionality during SIP call.
Is there any SIP client with these feature...i know WM has inbuilt SIP client...does it have these 2 features...its lot of work to enable it in..so just want to make sure beforehand
Regards,
ok pardon my noobiness above...so I configured the built in sip client and found it work just like regular call...and also it sucks compared to fring...there is more lag, sound breaking compared to fring(on phone data plan and even on wi-fi) and it misses incoming SIP calls too some times...
So is there any other solution?
You can try this one ..I've had good results with it.
Agreet agephone
http://www.ageet.com/us/products-agephone-mobile.htm
TechnoHippie said:
You can try this one ..I've had good results with it.
Agreet agephone
http://www.ageet.com/us/products-agephone-mobile.htm
Click to expand...
Click to collapse
Thanks for the suggestion...for now I've installed PortSIP and its working pretty good...I'll give that one a try also and compare..
Mafioso said:
Thanks for the suggestion...for now I've installed PortSIP and its working pretty good...I'll give that one a try also and compare..
Click to expand...
Click to collapse
Well I've tried both PortSIP and Agephone....both are good clients but none of them are able are as successful on T-mobile Edge network as Fring...it somehow manages to get adequate audio quality to get talks done...maybe it uses low bandwidth or something...The others have choppy/muffled voices in between
Is there any other client as successful as this? i've tried Nimbuzz, but it has the same problem as others..

[Q] Native VOIP support

Doesn't the GNEX have built-in VOIP support?
My old HTC Desire with Oxygen ROM (AOSP 2.3) had this.
I assumed ICS would have it too, but I don't find anywhere to enter the sip key and password....
No it does not.
Aww, that sux.
Does any of the roms offer this?
nxt said:
No it does not.
Click to expand...
Click to collapse
SIP accounts have been natively supported since Gingerbread.
In ICS go into phone -> Menu -> Settings -> Internet call settings.
You can set up your SIP account there and call settings (when to use, receive incoming, etc).
Hm, yeah thats what I thought too, but I don't see any "Internet call settings" in the setup menu on my GNEX 4.0.2.
madsere said:
Hm, yeah thats what I thought too, but I don't see any "Internet call settings" in the setup menu on my GNEX 4.0.2.
Click to expand...
Click to collapse
You need to go into the phone app, the menu, settings. It was moved out of the main settings page.
Sent from my Galaxy Nexus using xda premium
Ah that's where it's hiding. Actually it kinda makes sense. Thanks!
Anybody want to recommend a good SIP service and say a little about them? Currently I am using GrooVe IP + Google Voice but who knows what is going to happen now that Google has screwed them over..
I was recommended sipgate.com and terrasip.com by friends. Not much experience with them yet but I was told they are good.
From my own experience, avoid voipdiscount.com, awful call quality.
SipGate is not going to work for me, at least for the time being.
Thank you for your interest in sipgate!
Due to the increased volume of new registrations within the last few weeks, we have unfortunately ran out of free numbers in your area and we are currently not able to offer you any alternative free phone number at the moment.
We have ordered a new batch of numbers which will be made available very soon but we're afraid we can't give you an exact time frame for this yet. Follow us on twitter and we will let you know when it will be possible to sign-up for an account with us again.
We apologize for the inconvenience caused by this.
Click to expand...
Click to collapse
Edit: Terrasip looks more like a European service. Anyone had success with them in the US?
Ive been so spoiled on the whole Google Voice being free thing, I'm not so sure I'm willing to pay a minute fee for VOIP/SIP..
WiredPirate said:
Anybody want to recommend a good SIP service and say a little about them? Currently I am using GrooVe IP + Google Voice but who knows what is going to happen now that Google has screwed them over..
Click to expand...
Click to collapse
CallCentric
didlogic.com
Few people know about this provider because they mostly sell to other companies but their quality and rates is second to none. I pay $6/monthly and use the native sip feature. works like a charm.
Can anyone explain in a few words what is the difference between DID and SIP and VOIP.
I live in Thailand. I'd like to be able to let people call a number somewhere in Europe or US and have that redirect to my mobile here. Obviously with as low cost as possible.
I'd also like to be able to call people worldwide without spending an arm and a leg. That is what a SIP provider provides, correct?
I guess to get a number somewhere I need to find a DID provider able to provide the number in that country, right? And I'd still need some way to carry the call from the DID to me? Do I understand this right?
Unfortunately the international Internet connectivity from here isn't too good, so I thought perhaps it would be best to get a SIP provider here? Or does that not make any difference?
Sorry if this is going somewhat off topic, but since you guys seems to know this stuff and the OP (my) question has been answered I guess it's ok
Anyone know of a way to set it to only use internet calling when you are on wifi? It is pretty pointless for it to use 3g or LTE for internet calling...
Thanks
madsere said:
Can anyone explain in a few words what is the difference between DID and SIP and VOIP.
I live in Thailand. I'd like to be able to let people call a number somewhere in Europe or US and have that redirect to my mobile here. Obviously with as low cost as possible.
I'd also like to be able to call people worldwide without spending an arm and a leg. That is what a SIP provider provides, correct?
I guess to get a number somewhere I need to find a DID provider able to provide the number in that country, right? And I'd still need some way to carry the call from the DID to me? Do I understand this right?
Unfortunately the international Internet connectivity from here isn't too good, so I thought perhaps it would be best to get a SIP provider here? Or does that not make any difference?
Sorry if this is going somewhat off topic, but since you guys seems to know this stuff and the OP (my) question has been answered I guess it's ok
Click to expand...
Click to collapse
SIP is a protocol for VOIP (Session Initiation Protocol).
A DID is a direct inward dialing number (a telephone number).
Normally with a sip provider you don't get a DID. You receive calls at [email protected] This does not allow a standard phone to call you, just other SIP clients.
There are various services (and some SIP providers) that will allow/grant a DID to send a call to a SIP service. You can get a DID with Sipgate or use a service such as IPKall to get a DID and forward the calls placed to that DID to your VOIP/SIP provider (which will then ring your phone).
nomisunrider said:
Anyone know of a way to set it to only use internet calling when you are on wifi? It is pretty pointless for it to use 3g or LTE for internet calling...
Thanks
Click to expand...
Click to collapse
bump
10 char...

[Q] VoIP Problem

Hello everybody,
i have some Problems with my Galaxy Nexus GSM and Voice over IP. When i use Stock-ROM it works Perfectly. But every Custom-ROM have problems with it. When i get a call i can't hear the opposite. But he could hear me perfectly.
I use the voip support into the phone app from ICS. But also other apps don't work 'sipgate, sipdroid'
I except problems with my firewall because on stock it is working and on pc, too.
Now my question is, anybody else got problems like this?
regards
no ideas guys?
never tried out get voip work with custom rom?
klauhr said:
no ideas guys?
never tried out get voip work with custom rom?
Click to expand...
Click to collapse
I've had success with the integrated SIP client, but it has too many quirks and Verizon's network periodically blocks SIP by interfering with the Authorization headers (they strip them).
The problem is that it seems the Android SIP client interop testing was done against an Asterisk system and is very half-baked. It's like they got it working with one IP-PBX in the lab and figured it was good enough for wide use.
Regardless, that's not your question...
One-way audio is typically a firewall or NAT issue of some sort. What VoIP provider are you using? Is this your own IP-PBX system you're running at home? I do VoIP professionally so I can help.
I'm using sipgate it is an free provider here in Germany. If this is an Firewall/NAT problem my question is why it works with stock rom and pc and not with custom-Rom? It is confusing.
Im realy no expert in voip, my scheme is to have an landline number into my phone when i am at home. So i could receive calls on an landline. Because there are flatrates for landline but not for mobile here. The VoIP only must work over WLAN not with 3g.
thanks for the trouble
regards
I wish I could help more... I know these phones are finicky at times and do odd behavior. My wife's phone recently started to forget that it has an earpiece and microphone. Then, out of nowhere, it remembered that it did, but forgot it can vibrate. All of this survived multiple re-flashes, wipes, different ROMs... This phone is borderline terrible.
Are the ROMs you are working with all AOSP? 4.0.4? What version is the stock ROM? As a US user on a CDMA network, my ROM choices a very limited.
If you're comfortable with Wireshark, a perfect way to test thing would be packet captures. Compare the SIP headers between a working ROM and a failing ROM. If you find a real problem, you could get credit for a fix.

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