Radio ROM 1.48.00.10 Changelog (!) - 8525, TyTN, MDA Vario II, JasJam Software Upgradin

version No. 1.48.00.10
Description of Fixed Issue
1 3G cell is not reselected if 3G registration is rejected.
2 Radio will not send “No Carrier” when it receiving call disconnect with ECT successful facility
3 The device icon shows no service after flight on/off, but MO calls can still be performed.
4 Emergency call can not be dialed during conversation using a "Call Hold NOT supported" SIM.
5 After moving from restricted Cell to unrestricted Cell with location unregistered, location registration is performed with different RRC for LU and Attach.
6 Wrong handling of LU cause regarding to AC restricted -- Normal, Periodic, IMSI Attach -- all incorrected in different scenarios.
7 UE should respond to Paging of incoming call during AC restriction.
8【USIM】Free space of PLMN is updated with all "FF".
9 UE enters out-of-coverage state when UE registered with USIM of AC=0~9 and then enter cell that AC restriction is imposed on AC=0~10.
10 When PS call is originated after receiving Attach, LU, RAU Reject, it takes time to display connection error message.
11 UE always shows "Searching..." when UE camped in GSM and then goes to out-of coverage.
12 Answer Party B and end call from Party B at same time will cause DUT unable to switch back Party A properly (no sound).
13 [USIM]Phonebook synchronization is not performed when USIM is inserted in GSM terminal.
14 No-voice in uplink when 2nd call is canceled with End button during a speech call.
15 While UE perform cell selection procedure (for example, after call ended), UE should camp on a cell regardless of the AC restriction status of that cell.
16 Device keeps displaying call dialing when emergency call is dialed inside the shielding room.
17 Line2 can't MT call at all.
18 AT+CHLD=X has no response when no call exists.
19 Implement +ODEN which can be used by CE to add ECC defined by operator.
20 MS does not display SMS status reports that are undelivered.
21 Radio will not refresh SPN content when it receives Optus Dual SIM refresh command.
22 CS call can not be made after LAU failure with low layer failure and attempt counter greater then 4 and T3212 is running.
23 Uplink and Downlink will have no sound if CE issued AT+CHLD=1 to answer the incoming call.
24 3G : RAU with wrong Update Type in NMOI after CS call releases if RAI change during CS call.And this would cause call failed after that.
25 After flighting on/off, sometimes the device will not stay on the manually selected operator.

Related

[APP][UPDATE Aug 23,2010] PortGo(PortSIP CE) VoIP/SIP free softphone

PortGo for Windows Mobile 6/6.5 V1.3
PortGo is the newest free softphone application from PortSIP, it's built base on PortSIP VoIP SDK, allowing users to enjoy multimedia communications in a dynamic way.
PortGo works seamlessly with your internet connection – you can chat away with free calls and never worry about cost, time or distance. It including great features to help you stay in touch with friends, family and co-workers, share your thoughts and views and find the information you need. You can use it on your PDA
PortGo is a VOIP softphone that running on Windows Mobile 6.0/6.5, allows you to speak with any other softphone , any stand-alone IP-phone, or using Internet Telephony Service Provider (ITSP) with any traditional wired or mobile phone. It supports SIP and is fully inter-operable with most major VOIP vendors and ITSPs.
Standard Features
Support platforms: Windows Mobile 6/6.5
Support Device Screen: QVGA(320*240),VGA(640*480),WVGA(800*480)
Support servers: Cisco CallManager, Open SER, SER, Asterisk, Portaone, Radvision, Nortel, Rainbow, Avaya and other SIP Platforms.
Switch between earpiece and load-speaker(Does no support some devices, likes Samsung I708)
Audio call: G.711 aLaw/uLaw, GSM
Call transfer: Blind transfer
Call hold, mute speaker, mute microphone.
Audio record: record audio as wave file.
DTMF support: Send DTMF tone(RFC2833 and SIP INFO method), detect DTMF tone(RFC2833 and SIP INFO method).
Silent Ringtone
Audio conference
Acoustic Echo Cancellation
Automatic gain control
Comfort Noise Generation
Voice Activity Detector
STUN support
Bluetooth headset support
Outbound proxy server support
Jitter buffer
Customization SKIN,Default support:CIF(240*320),HVGA(240*400),WVGA(480*800)
Multiple Language support.User can add new language support.
Freeware
Todo:
1.Audio conference.
2.IM Support: SIMPLE(Presence, Subscribe, Pager message).
3.Support video call.
4.Audio record: record audio as MP3 file.
5.Support TLS/SRTP.
6.Customization Ringtone
7.Multiple sip accounts
Attention!
Having PortSIP installed is not enough to talk except over your local network. You have to obtain an account with any ITSP like Vonage or SipIptel.org, or install your own VoIP server.
If you have any advice, please let us know, we are happy to listen to.
Change log:
AUG 22, 2010 Version 1.3.8.18
1: Added set ptime options.
2: Added Acoustic Echo Cancellation(AEC),Noise Suppressor(ANS) options.
3: Support 2 line, audio conference.
4: Added MessCall notify
5: Added auto connect to Internet.
6: Add HVGA(240*400) skin.
7: Optimized call to PSTN sound quality.
8: Fixed some BUG.
Jun 17,2010
1: Reduce the voice delay.
2: Customization SKIN,Default support:CIF(240*320),WVGA(480*800).
3: Multiple Language support.User can add new language support by self.
4: Added Display name Multiple Language support.
5: change soft name to PortGo.
6: Support receive incoming call on Power suspend state.
7: Remove audio codec G.729,G723.1(Because the G.729 and G.723.1 codec is not free)
8: Fixed dial prefix BUG.
9: Agent name change to :"PortGo for Mobile"
Jul 8, 2009
1:Optimized sound quality.
2:Update SIP protocol stack.
3:Add button to remove account.
4:The number on the contact list auto Remove ()-, and space.
5:Fixed call out has the early media,the ringing sound comes through speaker,Now change the ringing sound comes through earpiece.
Jun 24,2009
1.When callnum is null, click call button will open "Call History".
2.The number on the contact list auto Remove ()- and space.
3: Set default port to 5060.
4.Disable "Auto screen off " .
5.Fixed "The application exit,but ringing tone continues" BUG.
6.Order account by last of user access.
7.Multiple tone files for the DTMF key presses.
8.Added show call failure message.
Jun 15,2009
1.Add windows contact support.
2.Disable Suspend when call is active(quickly drain the batteries).
3.Auto screen off when call is active.
4.Dial prefix support.
5.Command line support.
6.Long press(more then 2sec) of "0" changing to "+", when call num is empty.
7.Check for update.
8.Agent name change to :"PortSIP CE for Mobile"
9.reconnect or auto reconnect
10.fix some bugs.
May 20,2009
1.Add support DisplayName.
2.Fix AuthName bug.
3.Smaller memory requirements
April 24,2009
PortSIP CE Beta1 release
Download:
PortGo for Windows Mobile (Freeware)
http://www.portsip.com/downloads/portgo/PortGo_m.cab
PortGo Pro for Windows Mobile(15 days trial )
http://www.portsip.com/downloads/portgo/PortGo_Pro_m.cab
Multiple Language FAQ:
http://forum.xda-developers.com/showpost.php?p=6866807&postcount=364
last updated: Jun 17,2010
Support Device List:
HTC Touch Diamond
HTC Touch Diamond2
HTC Touch HD
HTC Touch HD2
HTC Touch 3G
HTC TyTN II
HTC Touch Diamond 2(looks kinda fuzzy)
HTC sprint mogul/titan 6800
Samsung Omnia i900
If PortGo can support/nonsupport you device, please tell to me, Help create support list,thank you!
Report Type :
Device Name | OS version | Support | False Reasons
sample:
HTC Touch Diamond WM6.1 YES
O2 Atom Exec WM5 NO Earpiece false
Settings
Hi I'm trying to use this software...Could you please help me with the settings?
I've created an account with iptel.org but can i use this account to make calls for free?
how does it work??
which proxy server and port do i have to use on the settings?
With the iptel.org service only pure VoIP calls can be made; iptel.org does not sell minutes into PSTN.
Sleeping while working !
Dear MaxDan,
Thank you very much for the time that you spent to find this and sharing with us !
i have tested this on TyTN ii and it succeeded ! I think this softphone is nice than the AgePhone mobile2, it has more codecs including G729 in a freeware, which are not available in AgePhone mobile2 after ($35).
There are few small bugs, which could be taken care in next update:
*It forgets the toggle setting for Earpiece / Speaker (some times toggles by itself)
*It also have a sleeping habit, when the screen goes off or the phone goes into standby mode, you can't hear your call, but its working in background.
Rest i would like to say its great software till date...
Thanks & Best Regards
Kip
Dear kipciaan,
Thank you for the use of PortSIP CE.
Question 1:
*It forgets the toggle setting for Earpiece / Speaker (some times toggles by itself)
Did you mean when selected "Use Earpiece", the phone still play ring tone on speaker when the call is incoming? This is our design.
*It also have a sleeping habit, when the screen goes off or the phone goes into standby mode, you can't hear your call, but its working in background.
This is a trouble, Because i can't disable sleep(Power will shortage), and i don't know connect to internet use WIFI or 3G or EDGE, which device must be activating. We will choose a compromise solution to fix this BUG at last version.
Best Regards!
Advance Settings
Dear MaxDan,
thanks for your reply!
* it toggle by it self :
when i generate CALL selecting "USE EARPIECE" mostly the sound comes from the EARPIECE but sometimes from SPEAKER (at back) on TyTN ii.
i think may be the file used for audio routing delay to execute or something... ('m not a technical pro_ )
there must be solution for the backlight and to hold the softphone running without shut off.
in the Menu i see the "OPTIONS" are hidden, don't know why?
*Please change the "History" button to "Contacts/PhoneBook" making others life simple!
#the "CALL History" can be viewed by going into menu.
waiting for the latest updates soon.
Best Regards.
Finally, a SIP tool. Thank you so much! Tow further enhacements:
-if possible in the future to allow configuration of a dial prefix, some VOIP providers require e.g 001 +phone number, if selected from Addressbook 001 is missing, so a possiblity to dial automatically 001+ addressbook number (or any other prefix) is welcome
-also multiple SIP accounts if possible, to switch between providers,
-how can I replace beep when a number is dialed with a DTMF tone
Thanks for the nice weekend present, and congrats for this very usefull tool
dear kipciaan,
* it toggle by it self
I will test this problem later.
there must be solution for the backlight and to hold the softphone running without shut off.
This BUG will fix shall precede all other.
the Menu "OPTIONS"
This is a reserved Menu. To implemented some specifically function.
"Contacts/PhoneBook"
We have to consider use windows mobile system contacts or PortSIP CE program self contacts. when implemented contacts,will change "History" button to "PhoneBook"
Best Regards.
Dear yo3gjc,
if possible in the future to allow configuration of a dial prefix, some VOIP providers require e.g 001 +phone number, if selected from Addressbook 001 is missing, so a possiblity to dial automatically 001+ addressbook number (or any other prefix) is welcome
We can add dial prefix future, setting at "Options" menu.
-also multiple SIP accounts if possible, to switch between providers
You can login out to change SIP accounts. We don't support use multiple SIP account at the same time.
how can I replace beep when a number is dialed with a DTMF tone
You can replace audio file at sound directory, 8000HZ mono 16bit wav.
i wasnt successful to get my VoIPBuster account registering
How can i use this software with voipBuster?
hello,
works fine with french freephonie sip provider.
Is it the same than pangolin ?
Can it be skinable?
thanks :.)
Using a VoIPBuster Account
Please try these settings!
Account name : your VoipBuster username
Password : your VoipBuster password
Proxy server : sip.voipbuster.com
SIP port : 5060
Under Advance tab:-
Domain
Auth Name:your VoipBuster username or voipnumber
User Domain: (leave it Blank)
Stun and Firewall OutBond Proxy:
Stunserver (option) : stun.voipbuster.com port 3478
Outbound proxy server : leave empty
You are requested to use G.711 codec for best results
Best Regsrds
Thanks a lot MaxDan, IMHO this is the biggest step ahead since X-pro, not to many apps around. Once again thanks for this initiative
MaxDan said:
Dear yo3gjc,
if possible in the future to allow configuration of a dial prefix, some VOIP providers require e.g 001 +phone number, if selected from Addressbook 001 is missing, so a possiblity to dial automatically 001+ addressbook number (or any other prefix) is welcome
We can add dial prefix future, setting at "Options" menu.
-also multiple SIP accounts if possible, to switch between providers
You can login out to change SIP accounts. We don't support use multiple SIP account at the same time.
how can I replace beep when a number is dialed with a DTMF tone
You can replace audio file at sound directory, 8000HZ mono 16bit wav.
Click to expand...
Click to collapse
so what else am I supposed to install on my IMate-8502 other than this software?
I have a SIP account from poivy.com.
Thanks
Handfree
This software works grate with my sip account .
Just 3 requests I have for next version if they are possible :
1) Use windows contacts to dial from the windows phonebook with our sip account.
2) Is it possible for this program to have an option to transfer the voice to bluetooth headset when there is a one connected to the phone ?
3) I also request for multiple sip accounts at the same time. The reason is my sip provider have a different proxy for incoming call and another one for outgoing with 2 different port numbers. Right now I can make calls no problem but I can't receive any calls because of the above reason .
If u can add several or at least 2 proxy at the same time, this software will be my Sipphone for ever.
Tnx
Thunder
hi Tempest69,
Is it the same than pangolin ?
Yes,It's same than pangolin. But pangolin is run at windows 2000/xp/vista, Pangolin is freeware too.
Can it be skinable?
We only support Customization Softphone for company current.
Freeware isn't support skinable.
hi thunder6290.
1.Use windows contact
we will add it at next version(Beta2).
2.bluetooth headset
3.multiple sip accounts
I has add it to todo, but it will add to PortSIP CE latter. maybe Beta3 or Beta4.
I has update todo, you can find it at Page1.
Anyone tried this with gizmo5? If so what settings did you use?
finally a newish voip software for WM6 and it's free.
Will try it out tonight after I get home from work.

Unknown Caller

Hello I have a HTC Fuze currently running Titanium 28005.
I am wondering if anyone else has had this problem also. When someone calls me from a cell phone if they are in my contacts or not it comes up as Unknown Caller but if someone calls me from a land line it will come up as it should.
I have tried installing Titanium 28005 and 21877 , Standard 28005 and 21877 also the Leo EnergyROM and still no matter what I do it comes up as Unknown Caller.
Everything I have tried to do:
Install different Roms
Hard Reset
Soft Reset
Deleting all of my contacts and putting them in again. With or without area codes and 1 with area codes.
Tried NRGZ28 Tachi Dialer Disabler.
Tried searching xda-developers with google.
I am sorry if I am being a "noob" or something and if I seem like I am bableing on I am just trying to give as much info as I can.
Caller Display falls into two major categories:
Landline
The majority of landline service providers include Caller Name Display (CallID) as part of their service offering. When someone wishes to keep their number private, they must pay a fee. Note that some landline service providers split the service offering as follows:
Display telephone number (usually free)
Display telephone number and name (enhancement, fee)
Cellular
As cellular telephone numbers are private by default, most subscribers (you) must be pay a fee to provide the number. Many cellular service provider "include" the feature as part of a basic (voicemail, sms, etc.) cellular feature package.
The majority of cellular service providers do not include Caller Name Display (CallID) as part of their service offering. The cellular subscriber must pay a fee to receive the telephone number of the calling party.
What this means:
When you call a landline number, your number would be displayed providing the providing the following rules are met:
You have not requested privacy.
The caller cellular service provider offers the number (and name) to other telco networks.
The landline customer is able to receive number (and name) display via their landline service provider.
You are paying for the Caller Name Display feature.
Your device is capable of processing number (and name) information.
When you call a cellular number, your number would be displayed providing the other cellular customer is paying for the service to display telephone numbers (and names).
You have not requested privacy.
The caller cellular service provider offers the number (and name) to other telco networks.
You are paying for the Caller Name Display feature.
Your device is capable of processing number (and name) information.
When you receive a call from a landline number, the number (and name) will appear on your display providing the following rules are met:
Caller has not requested privacy.
The caller landline service provider offers the number (and name) to other telco networks.
Your cellular service provider offers CallerID - some providers only offer number display, other offer number and name (CNAP).
You are paying for the Caller Name Display feature.
Your device is capable of processing number (and name) information.
When you receive a call from a cellular number, the number (and name) will appear on your display providing the following rules are met:
Caller has not requested privacy.
The caller cellular service provider offers the number (and name) to other telco networks.
Your cellular service provider offers CallerID - some providers only offer number display, other offer number and name (CNAP).
You are paying for the Caller Name Display feature.
Your device is capable of processing number (and name) information.
HTH,
Update: What it is, is that when ever a Verizon cell phone calls me it comes up as unknown caller. As an example when my girlfriend calls me (AT&T Phone) it comes up with her contact info. When ever a friend with Verizon calls me it comes up as Unknown Caller. I have tested this with all of my friends that are on Verizon My girlfriend has the same problem on her LG Vu that still stock nothing has been done to the phone, but my mothers phone works when a Verizon phone calls her. I have not changed anything on my bill or anything in the past 2 years only thing I have done is change phones.
Everything was working just fine in till Saturday I started getting the unknown caller problem. My friends have also mentioned that when they call me they hear a wired noise like a computer trying to connect to the internet thorough dial up then it starts to ring.
I contacted AT&T today and they are looking into the problem. It might have something to do with them merging with Centennial Wireless
I will keep people posted if they are interested in knowing what come out of it. And if someone can throw in there two cents on what they may think it to be be my guess I am willing to try anything. I hate now knowing who is calling.
Next time you speak with AT&T, ask them to "repush" your SIM profile as well to eliminate the possibility of profile corruption.
HTH,
I received a call from a Verizon line yesterday, this morning actually a little after midnight. Showed up. I am on AT&T.
Update: I just Re-Flashed the orignal HTC Rom that they provide on there website
(HTC FUZE RUU Raphael Cingular US 5.11.502.2 Radio Sign Raphael 52.64c.25.35 1.14.25.35 Ship)
And had my friend call me and everything came up as it should so it might be the rom I flashed to my phone witch was Phoenix 2 28005 11-27-2009
altho att might have done something with out contact me so i am going to re flash the Phoenix 2 28005 11-27-2009 and check
Update: Ok flashed Phoenix 2 28005 11-27-2009 and everything is working as it should. I don't know if AT&T did something or not but now when someone on Verizon calls me it comes up as it should.
when it comes unknown caller, is it display their number just below unknown caller?

Internet connection and phone calling

Hi All,
I've updated my Titan to TANGO and new firmware (and radio fw), but I have the same problem I had with Mango:
Data (internet/mail/tethering (shared internet connection) does NOT work if a phone call is on!
IE: I used tethering with my PC, if a phone call come no data connection, when I closed the call data connection restart!
This happens with EDGE and most of the times even in HSDPA... with my previous HD2 (WP6.5) I could use interner with my PC (in tethering) having a call in the same time even with EDGE...
Is this happenng to someone else?
PS: I inserted manually APN to be sure it is correct... no changes
BYE
I am having the same thing happen to, when browsing the web I cannot receive phone call.
Stock rom: OS-7.10.8107.79
Firmware- 1600.2200.20501.401
Radio-16.23.06.10_16.32.00.23U
Boot loader-2.5.160015.3(137079)
On which roms are you guys? Stock or customs?
f.
You know that it is totally normal that you can´t use data if a phone call is active ???
hi,
this happend on Mango and happens with Tango too (official HTC rom)
When I receive a phone-call, connection data gets out and NO internet connectio!. When the call finish the internet restarts again.
This do not happen on DEll Venue PRo with official Tango: internet keeps going when i speak by phone, as on my old HD2 with WM6.5
HELP!
I often use internat-sharing (tethering) and if I receive a call I can not answer if I qant to keep the internet connection!
THANKS
drky said:
I am having the same thing happen to, when browsing the web I cannot receive phone call.
Stock rom: OS-7.10.8107.79
Firmware- 1600.2200.20501.401
Radio-16.23.06.10_16.32.00.23U
Boot loader-2.5.160015.3(137079)
Click to expand...
Click to collapse
Hi have the opposite problem:
when browsing the web I receive a phone-call, but if I answer internet stops (under EDGE, HSDPA...) and restarts when I finisch the call
It is not possible to have both data and voice in EDGE mode. That's the limitation of GSM radio on that mode.
You need to be in 3G mode in order to do so. Old WM phones can automatically disconnection data if there is call come in and reconnect after you handup.
zorroz said:
Hi have the opposite problem:
when browsing the web I receive a phone-call, but if I answer internet stops (under EDGE, HSDPA...) and restarts when I finisch the call
Click to expand...
Click to collapse
I preferred your your problem, (although I don't think it's a problem) I don't mine receiving phone call and losing data.
foxbat121 said:
It is not possible to have both data and voice in EDGE mode. That's the limitation of GSM radio on that mode.
You need to be in 3G mode in order to do so. Old WM phones can automatically disconnection data if there is call come in and reconnect after you handup.
Click to expand...
Click to collapse
No, on my Dell Venue Pro under EDGE I can surfing web and syncronize email while I'm calling!
I do it, but I don't with my Titan
After 1000 tests with 4 sim I can say:
- on my Titan I can call and have connection only under HSDPA if I manually set APN
- on other devices (Dell Venue Pro with WP7.5 Tango + HD2 WM6.5) I can call and surf the web with EDGE 3G HSDPA
is it only on my device?
I know that EDGE is not very often present, but it happens in Italy... and in some places where I work!
PLS, if someone could make a test... thanks in advance
zorroz said:
No, on my Dell Venue Pro under EDGE I can surfing web and syncronize email while I'm calling!
I do it, but I don't with my Titan
Click to expand...
Click to collapse
I don't know how your DVP does it but do a search and you will learn that it is not possible for EDGE connection to serve both voice and data. It could be that you are really connected to 3G but your DVP is reporting EDGE. Or your DVP has two separate radio, one for data, one for voice.
foxbat121 said:
I don't know how your DVP does it but do a search and you will learn that it is not possible for EDGE connection to serve both voice and data. It could be that you are really connected to 3G but your DVP is reporting EDGE. Or your DVP has two separate radio, one for data, one for voice.
Click to expand...
Click to collapse
Thanks for your suggestions!
I searched in the web, and maybe I found out the answer from
http://www.techrepublic.com/forum/q...a-gets-disconnected-when-call-comes-on-mobile :
EDGE (and GPRS) are capable of supporting simultaneous voice and data ONLY if it is a Class C device.
Most phones these days are Class B (such as iPhone).
Class B support voice calls while in data mode (data disconnects when a call comes in) and data is "attached" while the phone is idle. Some Class B devices are configured so that the data takes priority (iPhone). Thus an iPhone will not take a call when an active edge data transfer is ongoing..it goes to voicemail.
Class A devices are either in data mode or voice mode (but never both).
even supported here:
http://reviews.cnet.com/8301-19512_7-10115034-233.html
"GPRS network 130 can be designed to operate in three network operation modes (NOM1, NOM2 and NOM3). A network operation modes of a GPRS network is indicated by a parameter in system information messages transmitted within a cell. The system information messages dictates a MS where to listen for paging messages and how signal towards the network. The network operation mode represents the capabilities of the GPRS network. In a NOM1 network, a MS can receive pages from a circuit switched domain (voice call) when engaged in a data call. The MS can suspend the data call or take both simultaneously, depending on the ability of the MS, In a NOM2 network, a MS may not received pages from a circuit switched domain when engaged in a data call, since the MS is receiving data and is not listening to a paging channel In a NOM3 network, a MS can monitor pages for a circuit switched network while received data and vise versa. "
This means that HTC Titan is not a Class C device...
foxbat121 said:
I don't know how your DVP does it but do a search and you will learn that it is not possible for EDGE connection to serve both voice and data. It could be that you are really connected to 3G but your DVP is reporting EDGE. Or your DVP has two separate radio, one for data, one for voice.
Click to expand...
Click to collapse
drky said:
I am having the same thing happen to, when browsing the web I cannot receive phone call.
Stock rom: OS-7.10.8107.79
Firmware- 1600.2200.20501.401
Radio-16.23.06.10_16.32.00.23U
Boot loader-2.5.160015.3(137079)
Click to expand...
Click to collapse
Your answer here:
http://reviews.cnet.com/8301-19512_7-10115034-233.html
We previously noted that the iPhone may miss calls while sleeping (locked). Some users are now reporting what seems to be a significant issue where their iPhones cannot receive incoming calls while transferring EDGE data. Here, in a nutshell, is what appears to be happening:
The iPhone cannot simultaneously use EDGE and voice services. That is, if you are on a call, you cannot concurrently access EDGE-data functions. In addition, as documented in Knowledge Base article #305711:
"While iPhone is actively transferring data over EDGEâ??downloading a webpage, for exampleâ??you may not be able to receive calls. Incoming calls may go to voicemail."
The "may not be able to receive calls" portion stems from the fact that there are two types of EDGE network types, NOM1 and NOM2. When your iPhone is connected to a NOM1 network, the data transmission will generally be interrupted, and the incoming call allowed to come through. When your iPhone is connected to a NOM2 network, however, the EDGE data transfer process cannot be interrupted, and the call will generally go to voicemail.
Fortunately, there is a way to check whether you are connected to a NOM1 or NOM2 network. First, put your iPhone in field test mode by accessing the Phone application, tapping Keypad, then entering *3001#12345#* and pressing Call.
Tap GPRS Information and look at the entry next to nom. It will be either 1 or 2. If you see a 1, you'll likely be able to receive a call while transferring data. If you see a 2, you likely won't.
Unfortunately, it appears that NOM2 is much more prevalent on AT&T's data network. In fact, we've yet to see our iPhone connect to a NOM1 network. As such, we can't even state with certainty that the iPhone supports NOM1 at all.
"GPRS network 130 can be designed to operate in three network operation modes (NOM1, NOM2 and NOM3). A network operation modes of a GPRS network is indicated by a parameter in system information messages transmitted within a cell. The system information messages dictates a MS where to listen for paging messages and how signal towards the network. The network operation mode represents the capabilities of the GPRS network. In a NOM1 network, a MS can receive pages from a circuit switched domain (voice call) when engaged in a data call. The MS can suspend the data call or take both simultaneously, depending on the ability of the MS, In a NOM2 network, a MS may not received pages from a circuit switched domain when engaged in a data call, since the MS is receiving data and is not listening to a paging channel In a NOM3 network, a MS can monitor pages for a circuit switched network while received data and vise versa. "
zorroz said:
This means that HTC Titan is not a Class C device...
Click to expand...
Click to collapse
As far as I know, there isn't any class C devices on the market. Maybe your DVP is an exception but I highly doubt it.

Call forwarding on PHOTON 4G

How do i activate call forwarding on my PHOTON 4G?
i am on non-sprint Network in India.
Thanks
It is depends on your carrier not you phone. IIRC there is no app or setting that will reroute a call that you phone receives. It is rerouted or forwarded by your carrier.
Call Forwarding
this is from wikipedia.
I am on BSNL 3G network. it works for me. (I could not see any call divert menu on my Photon 4G)
Forward service Activate Cancel & Deregister Cancel & Retain Status Reestablish Unconditional forwarding:
all calls *21*[phone number]# ##21# #21# *#21# *21#
Conditional forwarding:
if busy *67*[phone number]# ##67# #67# *#67# *67#
if not answered *61*[phone number]# ##61# #61# *#61# *61#
if out of reach *62*[phone number]# ##62# #62# *#62# *62#
all forwards *002*[phone number]# ##002# #002# *#002# *002#
all conditional forwards *004*[phone number]# ##004# #004# *#004# *004#
If the prefix to the forwarding command is "**" (instead of the usual "*"), then the phone number in that command is registered in the network. If after that the forwarding is deactivated using a command with a single "#", then later it will be possible to re-activate this forwarding again with a simple "*" command without a phone number in it. The forwarding will be re-activated to the number registered in the network. For example, if one uses the out-of-reach code in a forwarding command:
**62*7035551212#
and after that one deactivates the forwarding:
#62#
then later it will be possible to re-activate the out-of-reach forwarding without specifying a number:
*62#
After the above command, all calls made to the phone, while it is out of reach, will be forwarded to 7035551212. It is possible to activate the feature to a number other than the registered number, whilst still retaining the registered number for later use. For example, issuing the command:
*62*7185551212#
will result in calls being forwarded to 7185551212 (and not to the registered number 7035551212). However, if later a command is issued:
*62#
then the calls will again be forwarded to the registered number 7035551212 (and not to the number from the previous forwarding command 7185551212).
Additionally, in GSM networks, such as T-Mobile and AT&T Mobility in USA, and all mobile networks in EU, it is possible to set the number of seconds the phone will ring before forwarding. This is specified by inserting "*SC*XX" prior to the final "#" of the forwarding command, where "SC" is a service type code (11 for voice, 25 for data, 13 for fax), and "XX" is the number of seconds in increments of 5 seconds. If "SC" is omitted (just "**XX") then by default all service types will be forwarded. For example, forwarding on no-answer can be set with:
*61*[phone number]**[seconds]#
In some networks there may be a limit of not more than 30 seconds before forwarding (i.e. "XX" can only be 5, 10, 15, 20, 25 or 30; all greater values, like 45 and 60, will result in the forwarding command being rejected and an error message returned).
Thanx portman.. alas m not on BSNL network
Sent from my MB855 using xda app-developers app
Call forward is a network service. Photon normally do not allow it to activate outside sprint. may be they blocked it in the os level. But you can do it by putting the Sim in another phone and activate call forwarding and put the sim back in photon and it will work. Photon just dont allow to activate it but happy with already activated forwards!.
Sent from my MB855 using xda app-developers app
Great!"
Thnx
Sent from my MB855 using xda app-developers app

[Q] How to insert synthetic voice stream into GSM voice connection

First of all, I back your pardon for my very basic english.
In Germany I started few weeks ago a project for the development of a new android app to improve emergency calls (112 in the EU,112 and 110 in Germany) of speech handicapped persons using mobile phones. Aim is to send at the beginnig of a GSM voice connection a synthetic voice stream with the GPS position data of the Android mobile phone, for example : "This is an emergency call. Please prepare to receive the position data of the caller. Longitude is five one comma one two three four. Latitude is six comma one two three four. The data will be repeated. Longitude is five one comma one two three four. Latitude is six comma one two three four. Voice channel is open now.". During the GSM connection there should be inserted simple synthetic answers like "Yes", "No" or "Please repeat" by pressing a button on the display. This way is necesseray because there is no other incomming channel with the PSAPs in Germany to receive GPS positioning data. There is no way to receive SMS nor to enable a data connection between mobile phone and PSAP. For voice communication the GSM should be used only, no VoIP connections.
As I learned during the project there is no official APK or SDK for developers to get access of the GSM voice stream into the mobile.
Does anybody have an idea to solve this problem ? Is there an indirect way via the microphone, perhaps ?
Kind regards
RG

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