Database Develop

How To Disable the noise cancellation

Yes, the title says it.
Anybody an idea how to disable it?

Why would you want to disable it? Just curious.

Yes, easy to answering:
http://forum.xda-developers.com/showthread.php?t=700018
It's a real annoying problem by making calls with plugged in car radio or headset.
(Look at the Link)

It was recently posted in the general section that if you edit /system/build.prop from "media.a1026.enableA1026=1" to "media.a1026.enableA1026=0" it should disable the noice cancelling mic. I have not tried it myself yet but it should work. You need to re-edit the file if you want to re-enable the mic.

Thank you very very much.
This works!!
Now I can place calls in my car, plugged in to the AUX Jack and others can hear what I say.
I haven't found it yet in the forum...

pmadej15 said:
It was recently posted in the general section that if you edit /system/build.prop from "media.a1026.enableA1026=1" to "media.a1026.enableA1026=0" it should disable the noice cancelling mic. I have not tried it myself yet but it should work. You need to re-edit the file if you want to re-enable the mic.
Click to expand...
Click to collapse
Hi, How exactly do you do that? I am using a Samsung Galaxy S. Please help.

Your phone must be rooted.
Open root explorer, navigate to /system/
Mount your System r/w and open the build.prop with a long press and choose "open in text editor"
Then scroll to the correct line, delete the "1" and change it to an "0".
Then save and exit and don't forget to reboot your phone.

I have an Android G1 devphone in my possession and the /system/build.prop doesn't contain that string.
Any ideas on turning off the noise canceling mic?
Thanks!
/system/build.prop
Code:
# begin build properties
# autogenerated by buildinfo.sh
ro.build.id=CRC1
ro.build.display.id=dream_devphone-userdebug 1.5 CRC1 150275 test-keys
ro.build.version.incremental=150275
ro.build.version.sdk=3
ro.build.version.release=1.5
ro.build.date=Wed Jul 1 13:38:12 PDT 2009
ro.build.date.utc=1246480692
ro.build.type=userdebug
ro.build.user=android-build
ro.build.host=apa3.mtv.corp.google.com
ro.build.tags=test-keys
ro.product.model=Android Dev Phone 1
ro.product.brand=android-devphone1
ro.product.name=dream_devphone
ro.product.device=dream
ro.product.board=trout
ro.product.manufacturer=HTC
ro.product.locale.language=en
ro.product.locale.region=US
ro.board.platform=msm7k
# ro.build.product is obsolete; use ro.product.device
ro.build.product=dream
# Do not try to parse ro.build.description or .fingerprint
ro.build.description=dream_devphone-userdebug 1.5 CRC1 150275 adp,test-keys
ro.build.fingerprint=android-devphone1/dream_devphone/dream/trout:1.5/CRC1/150275:userdebug/adp,test-keys
# end build properties
#
# system.prop for dream
#
rild.libpath=/system/lib/libhtc_ril.so
wifi.interface = tiwlan0
# Time between scans in seconds. Keep it high to minimize battery drain.
# This only affects the case in which there are remembered access points,
# but none are in range.
wifi.supplicant_scan_interval = 45
# density in DPI of the LCD of this board. This is used to scale the UI
# appropriately. If this property is not defined, the default value is 160 dpi.
ro.sf.lcd_density = 160
#
# ADDITIONAL_BUILD_PROPERTIES
#
ro.config.notification_sound=F1_New_SMS.ogg
ro.com.google.locationfeatures=1
ro.com.android.wifi-watchlist=GoogleGuest
ro.com.android.dateformat=MM-dd-yyyy
ro.com.android.dataroaming=true
ro.com.google.clientidbase=google
ro.ril.hsxpa=1
ro.ril.gprsclass=10
ro.setupwizard.mode=OPTIONAL
ro.url.legal=http://www.google.com/intl/%s/mobile/android/android-dev-phone-legal.html
ro.url.legal.android_privacy=http://www.google.com/intl/%s/mobile/android/android-dev-phone-privacy.html
keyguard.no_require_sim=true
ro.config.sync=yes
net.bt.name=Android
dalvik.vm.stack-trace-file=/data/anr/traces.txt

G1 doesn't have such mic, I believe.

Jack_R1 said:
G1 doesn't have such mic, I believe.
Click to expand...
Click to collapse
Oh really? It's the best hypothesis that I've got.
I'm trying to make an app that is a dumb microphone that records all background noise (to be processed later) and the phone seems to ignore anything that isn't my voice. Even when I blast music and stick the phone next to the speaker, the phone records the music in very very low volume. Same thing happens if I try the Voice Search app (can tell volume based on the amplitude graph shown) and if I record video. All audio but voice is either ignored or hush hush in videos.
I can only think that it is a noise canceling mic destroying the background noise.
Any ideas?

HW "voice recognition" processing that doesn't use second mic, most probably.
I've checked the net, and saw no mention of G1 noise cancellation capabilities anywhere.

Jack_R1 said:
HW "voice recognition" processing that doesn't use second mic, most probably.
I've checked the net, and saw no mention of G1 noise cancellation capabilities anywhere.
Click to expand...
Click to collapse
a G1 definitely does not have a second noise cancellation mic

this works
this works. I have an n1 and did this quick mod and now I can talk and drive without any problems. I have my phone hooked up via headphone jack to the stereo and it sits about 2 feet away from me.
Thanks for help guys !

pmadej15 said:
It was recently posted in the general section that if you edit /system/build.prop from "media.a1026.enableA1026=1" to "media.a1026.enableA1026=0" it should disable the noice cancelling mic. I have not tried it myself yet but it should work. You need to re-edit the file if you want to re-enable the mic.
Click to expand...
Click to collapse
How do I get permission to edit the file? Please

Root, execute "adb remount".

pmadej15 said:
It was recently posted in the general section that if you edit /system/build.prop from "media.a1026.enableA1026=1" to "media.a1026.enableA1026=0" it should disable the noice cancelling mic. I have not tried it myself yet but it should work. You need to re-edit the file if you want to re-enable the mic.
Click to expand...
Click to collapse
Hi pmadej. I couldn't find it on my SGS2. And I even try to manually type in the line, but with no hope. Do you know how to disable it on SGS2? Thanks in advance.

hello.
can anyone help me how to do that on lenovo a706?
i have low level mic problem when calling..and i thing is noise cansellation.
please help me on this..
thank you

People, and how do you not to disable noise cancelation, but fix it to do less noise canceling? Is there a mod for Nexus 5?

On my build.prop I can't find "media.a1026.enableA1026=1". If I add this line to my build,prop it will work?

i also dont have that part of code on ot-995 , what to do developers ?

[SBF]D3 2.3.4 SBF leaked by China Forums

So with a few hours of Googling and some searching, me and a few others on IRC have found a mysterious D3 2.3.4 SBF. Of the time of writing, we're still downloading as the download is incredibly slow. Hopefully some of the people that are in the IRC will post mirrors on this thread as soon as they can get it uploaded.
http://www.diypda.com/zftp.action.p...763241688&md=71caf39d4e9a98681331bd16f2caa3f2
Download it at the very bottom of the thread. Mirrors will be up shortly. IRC people, please post mirrors if you have any.
MIRROR!!
Thanks to rigwig, we now have a mirror. Please thank him
http://www.multiupload.com/HKMICRBC6J
IMPORTANT!!:
DO NOT BY ANY MEANS FLASH THIS. THIS IS SIGNED WITH TEST KEYS, NOT RETAIL. FLASHING THIS WILL GUARENTEE A BRICK
here's your final warning message.
[13:25] whaatt: can we use rsdlite to flash this sbf?
[13:25] Adam: DO NOT FLASH THIS
[13:25] Adam: stop thinking about it
[13:25] Adam: stop thinking about thinking about it
[13:25] whaatt: lol
[13:25] whaatt: then what to do with it?
[13:25] royolpunk: DONT FLASH IT

downloading now. I'll have mirror up in a few mins
Edit:
Here is the Multiupload mirror
http://www.multiupload.com/HKMICRBC6J

*prays its real*
Time for some epic rooting methods!

This is as real as it gets. DOWNLOAD SPEED WILL BE 50KBps MAX.
Edit:
So we've got a few people on the IRC downloading away, one is at 50%. it'll take about another hour before we get everything cleared up. I'll try to update this thread ASAP.
p3 has also just joined us on IRC. we're doing our best

Any thoughts on this giving us root?
Sent from my DROID3 using XDA App

saintlikesniper said:
Any thoughts on this giving us root?
Sent from my DROID3 using XDA App
Click to expand...
Click to collapse
We're hoping to find vulnerabilities and holes in this Chinese SBF file, and see what we can do from there in terms of getting root.

we're hoping that this is a complete image of the phone (which it should be, considering it's at 222MB. we're going to go through the SBF step by step and look for more exploits that we can use to root this phone.
edit:
whaatt you ninja'd me -____-

This is /system/build.prop:
Code:
# begin build properties
# autogenerated by buildinfo.sh
ro.build.id=5.5.1-1_GC-35
ro.build.display.id=cdma_solana-userdebug 2.3.4 5.5.1-1_GC-35 5.5.1-1_GC-35 test-keys
ro.build.version.incremental=5.5.1-1_GC-35
ro.build.version.sdk=10
ro.build.version.codename=REL
ro.build.version.release=2.3.4
ro.build.date=Thu Jun 9 04:21:36 CDT 2011
ro.build.date.utc=1307611296
ro.build.type=userdebug
ro.build.user=hudsoncm
ro.build.host=il93lnxdroid48
ro.build.tags=test-keys
ro.product.model=XT883
ro.product.brand=moto
ro.product.name=SLNCT
ro.product.device=cdma_solana
ro.product.board=solana
ro.product.cpu.abi=armeabi-v7a
ro.product.cpu.abi2=armeabi
ro.product.manufacturer=motorola
ro.product.locale.language=zh
ro.product.locale.region=CN
ro.wifi.channels=
ro.board.platform=omap4
# ro.build.product is obsolete; use ro.product.device
ro.build.product=solana
# Do not try to parse ro.build.description or .fingerprint
ro.build.description=cdma_solana-userdebug 2.3.4 5.5.1-1_GC-35 5.5.1-1_GC-35 test-keys
ro.build.fingerprint=moto/SLNCT/cdma_solana:2.3.4/5.5.1-1_GC-35/5.5.1-1_GC-35:userdebug/test-keys
# end build properties
#
# system.prop for CDMA Droidwe
#
# rild.libpath=/system/lib/libril-moto-umts-1.so
rild.libpath=/system/lib/libmoto_ril.so
rild.libargs=-d /dev/ttyO0
ro.sf.lcd_density=240
ro.default_usb_mode=2
ro.mot.ril.danlist=611,*611,#611
persist.ril.ecclist=911,*911,#911,119,118,999,08,000,112,110
persist.ril.modem.mode =1
# This defines the max event window manager can
# handle in 1 s. We may adjust this # for performance
# reason later
windowsmgr.max_events_per_sec=55
# USB modes allowed for UI and switching
ro.ngp_available=1
ro.modem_available=0
# Default network type.
# 7 => GLOBAL.
ro.telephony.default_network=7
ro.com.google.clientid=android-motorola
ro.url.legal=http://www.google.com/intl/%s/mobile/android/basic/phone-legal.html
ro.url.legal.android_privacy=http://www.google.com/intl/%s/mobile/android/basic/privacy.html
ro.cdma.home.operator.numeric=46003
ro.cdma.home.operator.alpha=CT
ro.config.vc_call_vol_steps=7
ro.cdma.subscription=0
# wlan interface
wifi.interface = tiwlan0
softap.interface = tiap0
# Time between scans in seconds. Keep it high to minimize battery drain.
# This only affects the case in which there are remembered access points,
# but none are in range.
wifi.supplicant_scan_interval = 45
# Indicate carrier OTA SP number schema
# refer to frameworks/base/telephony/java/com/android/
# internal/telephony/cdma/CDMAPhone.java for the schema:
ro.cdma.otaspnumschema=SELC,1,80,99
# The OpenGL ES API level that is natively supported by this device.
# This is a 16.16 fixed point number
ro.opengles.version = 131072
# This is a high density device with more memory, so larger vm heaps for it.
dalvik.vm.heapsize=32m
# our HTML browser (not WAP browser ) does not need to specify UAProf URL
ro.mot.hw.uaprof=http://uaprof.vtext.com/mot/xt862/xt862.xml
# Enable HAC to also display Call Setting Menu Option.
ro.mot.hw.HAC=1
# Motorola, a19622, 12/28/2009, IKSHADOW-204 / Location Settings for android
# Property to set enabling state of the location privacy setting feature.
# Motorola, a19622, 03/16/2010, IKSHADOW-2141 / Disable feature 33067
#ro.mot.settings.locprivacy=1
# ERI feature
# 0 OFF
# 1 ON
ro.mot.eri=0
# Motorol,a6889c,02/Feb/2010,IKSHADOW-619
# Proximity Sensor feature
# 0 disable
# 1 enable
persist.mot.proximity.touch=1
# End IKSHADOW-619
# BEGIN Motorola, a22976, 25-Feb-2010, IKSHADOW-702, FID 31841
# VZW Requirement Specifications: NETWORK BASED PLUS CODE DIALING
# 0 disable
# 1 enable
ro.cdma.nbpcd=0
# END IKSHADOW-702
# BEGIN Motorola, a22976, 12-Mar-2010, IKSHADOW-1784, Power Up Tone
# Motorola, a22976, 12-Apr-2010, IKSHADOW-3298, Remove temporary power up tone
# persist.mot.powerup.tone=/system/media/bootup.ogg
# END IKSHADOW-1784
persist.mot.usb.mediasync = 1
# persist.usb.android_config = 1
# System property for SMC
ro.service.start.smc=1
ro.HorizontalBUA=true
#Proximity sensor debounce time
mot.proximity.delay=450
# disable touch below 60 pixels
mot.proximity.distance=60
#inphone calibration backup support
ro.mot.hw.calibratedImager=1
# BEGIN Motorola, qpmc46, 05-Jul-2010, IKMAIN-311
persist.ril.mux.noofchannels = 8
# END Motorola, qpmc46, 05-Jul-2010, IKMAIN-311
#BEGIN Motorola, nfjb73, 21-jun-2010,
# this is for feature 34386, other features for vzw global phone
# could use this property too. IKMAIN-732
#1 this is verizon wireless globle phone
#0 this is not verizon wireless globle phone
ro.mot.phonemode.vzwglobalphone=1
# END Motorola, nfjb73, 21-jun-2010, IKMAIN-732
#ro.mot.phonemode.numsignalbars=4
# BEGIN Motorola, nfjb73 23-jun-2010, IKMAIN-624
# this is for feature 33860, when phone is activated, the cdma mdn is written.
persist.radio.vzw.cdma.mdn=
# END Motorola, nfjb73
# BEGIN Motorola, qmfd78, 14-Jul-2010, IKMAIN-884
# GSM SMS traffic needs to be routed/sent through North America SMSC.
# This is part of the VZW World-phone Feature. SMS are always sent to a US SMSC and then injected
# into that network. This allows for billing and give the user the impression that the device is
# still in the US.
# ro.telephony.gsm-routes-us-smsc = 1
# END Motorola, qmfd78, 14-Jul-2010, IKMAIN-884
# BEGIN Motorola, a22600 qmfd78, 16-Jul-2010, IKDROIDPRO-423
# For north american cdma operators, this flag should be 1.
# It enables number mangling when nbpcd (plus code dialing) is not supported.
# This is part of the VZW World-phone Feature.
#ro.cdma.home.operator.isnan=1
# END Motorola, a22600 qmfd78, 16-Jul-2010, IKDROIDPRO-423
# EDM feature - IKDROIDPRO-381 - Start
ro.mot.dpmext = true
# EDM feature - IKDROIDPRO-381 -End
# BEGIN Moto,IKDROIDPRO-430, a17673
# Feature33857 on DroidWE, support DualMode switch, UMTS/CDMA
ro.mot.FTR.33857=true
# END IKDROIDPRO-430
# BEGIN Moto, IKSTABLETWO-1284, nfjb73
# without this property, the lock screen would take global phone as UMTS
# block user to operate before setup
keyguard.no_require_sim = true
# End Moto
# BEGIN Moto, IKSTABLETWO-689, nfjb73, for 34387,33857
# when phone in Global mode, FW need to know the last phone mode beore power off
# when phone power up, it would use the right mode to avoid switch
persist.radio.ap.phonetype=2
# END IKSTABLETWO-689
# Motorola, IKHALFMWK-3, w001091
# Feature 32711, deep sleep mode support
ro.mot.deep.sleep.supported=true
# BEGIN Moto, IKMAIN-14097, gdr487, 33531 flex on for Solana
# true feature enabled
# false feature disabled
ro.mot.fid.33531.keylock_ecm=true
# END IKMAIN-14097
# BEGIN Motorola, A24124, 01-Mar-2011, IKMAIN-13570
ro.mot.bindervm.config = 210
# END IKMAIN-13570
#BEGIN Motorola, w30471, 16-Mar-2011, IKSTABLEFIVE-1796 35691 - Set configurable parameters for Verizon products
mot.mms.con.timeout = 30000
mot.mms.socket.timeout = 120000
mot.mms.tr.vc.timeout = 55000
mot.mms.trm.timeout = 240000
mot.mms.abort.prog.lim = 90
mot.mms.retry.pic.size = 512000
mot.mms.retry.default = false
mot.mms.retry.timeout.1 = 300000
mot.mms.retry.timeout.2 = 300000
mot.mms.retry.timeout.3 = 300000
mot.mms.retry.timeout.4 = 0
# END IKSTABLEFIVE-1796
#BEGIN Motorola, w36256, 6-Apr-2011, IKSTABLEFOURV-6808
# add latin encoding type for sms in cdma mode, this is only for vzw
ro.cdma.sms.latin_encode=true
#END IKSTABLEFOURV-6808
#
# ADDITIONAL_BUILD_PROPERTIES
#
ro.kernel.android.ril=yes
persist.ril.baudrate=230400
persist.ril.mux.noofchannels=8
persist.ril.mux.ttydevice=/dev/ttyO0
persist.ril.modem.ttydevice=/dev/ttySPI0
persist.ril.features=0x30E
persist.ril.mux.retries=500
persist.ril.mux.sleep=2
persist.mot.mdm_panicd.nopanic=no
ro.config.ringtone=DROID3.ogg
ro.config.notification_sound=pizzicato.ogg
ro.config.notification_sound=OnTheHunt.ogg
ro.config.alarm_alert=Alarm_Classic.ogg
ro.default_usb_mode=0
ro.media.enc.aud.fileformat=qcp
ro.media.enc.aud.codec=qcelp
ro.media.enc.aud.bps=13300
ro.media.enc.aud.ch=1
ro.media.enc.aud.hz=8000
ro.telephony.call_ring.multiple=false
ro.telephony.call_ring.delay=3000
ro.setupwizard.enable_bypass=1
ro.cdma.homesystem=64,65,76,77,78,79,80,81,82,83
ro.cdma.data_retry_config=default_randomization=2000,0,0,120000,180000,540000,960000
ro.media.camcorder.1080p=mp4,h264,30,15000000,aac,128000,44100,2
ro.media.camcorder.720p=mp4,h264,30,10000000,aac,128000,44100,2
ro.media.camcorder.d1NTSC=mp4,h264,30,6000000,aac,128000,44100,2
ro.media.camcorder.vga=mp4,h264,30,4000000,aac,128000,44100,2
ro.media.camcorder.cif=mp4,h264,30,1500000,aac,128000,44100,2
ro.media.camcorder.qvga=mp4,h264,15,500000,aac,64000,44100,2
ro.media.camcorder.mms=3gp,h264,15,128000,amrnb,12200,8000,1
ro.media.camcorder.mmsres=qvga
ro.camcorder.zoom=true
ro.media.capture.maxres=8m
ro.media.capture.fast.fps=4
ro.media.capture.slow.fps=120
ro.media.capture.flash=led
ro.media.capture.flashMinV=3300000
ro.media.capture.torchIntensity=40
ro.media.capture.flashIntensity=70
ro.media.panorama.defres=3264x1840
ro.media.panorama.frameres=1280x720
ro.camera.dynamicFocus=true
ro.camcorder.videoModes=false
media.stagefright.enable-player=true
media.stagefright.enable-meta=true
media.stagefright.enable-scan=false
media.stagefright.enable-http=true
media.stagefright.enable-record=false
ro.media.camera.focal=3451.0,3451.0
ro.media.camera.principal=1632.0,1224.0
ro.media.camera.skew=0.0
ro.media.camera.distortion=0.0,0.0,0.0,0.0,0.0
ro.media.camera.calresolution=3264,2448
ro.mot.setuptype=2
ro.horizontalBUA=true
ro.horizontalVVM=true
ro.horizontalIM=true
ro.horizontalNGM=false
ro.horizontalVMM=false
ro.horizontalMOD=false
ro.horizontalVOD=false
ro.horizontalEmail=false
ro.mot.internalsdcard=1
dalvik.vm.lockprof.threshold=500
dalvik.vm.dexopt-flags=m=y
net.bt.name=Android
dalvik.vm.stack-trace-file=/data/anr/traces.txt
ro.mot.dpmext=true
ro.com.google.clientid=android-motorola
ro.url.legal=http://www.google.com/intl/%s/mobile/android/basic/phone-legal.html
ro.url.legal.android_privacy=http://www.google.com/intl/%s/mobile/android/basic/privacy.html
ro.mot.hw.uaprof=http://uaprof.motorola.com/phoneconfig/MotoMB200/profile/MotoMB200.rdf
debug.mot.hprofdump=1
ro.build.version.full=Blur_Version.55.11.35.XT883.ChinaTelecom.en.CN
ro.mot.hidden_keyboards=evfwd

This is a listing of all the files in /system: http://pastebin.com/XGXbDSNR

So if this becomes flashable and can we not inject busybox superuser and other stuff to gain root into the sbf
Sent from my DROID3 using XDA App

Yeah
Sent from my DROID3 using XDA Premium App

Man, this is exciting!! I hope this makes it easier for the devs!

This is a user debug build with limited su in xbin so adb shell has root but not on the phone.
These builds are easily rooted of course, but cannot be flashed on secured hardware at all.
The value of this file is more from the ability to examine the code group structure so we can look at the bootchain and CDT to get a better understanding how the phone works. This also appears to be for older P2 hardware revision, whereas the VZW released devices are P3.
That is another reason not to flash this file to our phones under any circumstances.
Thanks for digging this up and P3 and I will be taking a look at it as I qm sure many others will.
This does not mean we are any closer to root on stock hardware but is very exciting nonetheless.

I see...darn I thought we were getting pretty close to root
Sent from my DROID3 using XDA Premium App

royolpunk said:
we're hoping that this is a complete image of the phone (which it should be, considering it's at 222MB. we're going to go through the SBF step by step and look for more exploits that we can use to root this phone.
edit:
whaatt you ninja'd me -____-
Click to expand...
Click to collapse
I downloaded a file a while ago called "D3 system dump" want a link?

Nah we've already been through that
Sent from my DROID3 using XDA Premium App

royolpunk said:
Nah we've already been through that
Sent from my DROID3 using XDA Premium App
Click to expand...
Click to collapse
Gotcha, thanks. Trying any lead I can haha.
Sent from my DROID3 using XDA Premium App

Woo! I can't wait to see what the devs can do with this. Gonna be the first thing I check when I wake up in the morning
Sent from my DROID3 using XDA Premium App

Nothing... it was pretty disappointing actually.
Sent from my DROID3 using XDA App

Sent from my DROID3 using XDA Premium App

[GUIDE] Setup Your Own Asterisk Server With Google Voice on Amazon EC2

EDIT: THIS SETUP IS FOR PEOPLE WHO WANT A RELIABLE ALTERNATE TO USING MAIN STREAM SERVICES. THIS IS YOUR OWN SERVER -- AND YOU WILL BE THE ONLY ONE USING IT AS OPPOSED TO HUNDREDS OR THOUSANDS OF OTHER USERS. I'VE BEEN USING THIS FOR SIX MONTHS WITH NO ISSUES. BEFORE THIS I WAS USING PBXES.ORG AND THE SERVICE WAS 'OK' AND I MISSED SOME CALLS. I'VE HAD ZERO ISSUES WITH THIS SETUP.
A few people have asked me for a guide on howto setup Asterisk on an Amazon EC2 micro instance for their Nexus 4. Amazon offers a free micro instance for one year (new signups). Even after your first year it's pretty cheap to keep it running. Amazon's bandwidth will be faster than setting up Asterisk on your own home server (in most cases).
Once you have the server up and running you can easily configure your favorite SIP client on your Nexus 4 to enjoy free calling with your Google Voice account. This is particularly useful for people such as myself who are on the $30 T-Mobile plan.
As a bonus, I've also included the steps to setup a PPTP VPN.
This guide assumes that you've already setup your AWS account and figured out how to set the security group. You will need to open some ports (TCP: 22, 1723, 5060. UDP: 5060, 10000-20000)
Step 1. Goto: http://uec-images.ubuntu.com/releases/10.04/release/ and pick the t1.micro instance (ebs 64 bit) for the region that you setup in AWS. Launch this instance (there is a button) and get it working with the security group that you configured. After it's launched you need to setup an Elastic IP and associate it with the instance. After that go ahead and log into your new micro instance server. Once you get to this point, then you can continue with the guide. There are TONS of resources (including youtube videos) on how to get to this point. It's not rocket science.
Step 2. Setup firewall settings for pptpd and asterisk. Lucid also has firewall settings that need to be adjusted.
Code:
#Uncomplicated Firewall
sudo ufw enable
sudo ufw allow 22/tcp
sudo ufw allow 1723/tcp
sudo ufw allow 5060/tcp
sudo ufw allow 5060/udp
sudo ufw allow 10000:20000/udp
#check status
sudo ufw status
#edit /etc/default/ufw and enable forward policy
DEFAULT_FORWARD_POLICY="ACCEPT"
#edit /etc/ufw/sysctl.conf and uncomment
net/ipv4/ip_forward=1
#edit /etc/ufw/before.rules and add this after the header comments
---<BEGIN>--- (DON'T COPY THIS LINE)
# nat Table rules
*nat
:POSTROUTING ACCEPT [0:0]
# Forward traffic through eth0.
-A POSTROUTING -o eth0 -j MASQUERADE
# don't delete the 'COMMIT' line or these nat table rules won't be processed
COMMIT
---<END>--- (DON'T COPY THIS LINE)
#disable and enable to apply changes
sudo ufw disable && sudo ufw enable
Step 3. Recompile Kernel. The default kernel is set at 100HZ timing, this will give you HORRIBLE VOIP quality. The kernel needs to be recompiled to 1000HZ timing.
Code:
# Make yourself root
sudo su
# Update source list:
aptitude update
# Upgrade everything:
aptitude upgrade
# Install dependencies:
apt-get build-dep linux-image-$(uname -r)
apt-get build-dep linux
apt-get install fakeroot build-essential
apt-get install crash kexec-tools makedumpfile kernel-wedge
apt-get install libncurses5 libncurses5-dev
apt-get install libelf-dev asciidoc binutils-dev kernel-package
apt-get install git-core
cd /usr/src
git clone git://kernel.ubuntu.com/ubuntu/ubuntu-lucid.git
cd ubuntu*
git checkout --track -b ec2 origin/ec2
fakeroot debian/rules clean
fakeroot debian/rules editconfigs
# Configuration window should now appear, do the following:
Select YES
# Navigate to:
Processor type and features -> Timer frequency
# Select the 1000HZ frequency
Exit
Exit
Yes (Save)
#After saving and returning to prompt it may ask you to do it again for i386, select yes and repeat!
This next command will take about 7 hours to recompile the kernel. But, there is a shortcut. Amazon charges by the minute for each instance type that you use. I recommend shutting down your instance at this point and changing it to a m1 extra large instance type (this will cost you about 70 cents). This will increase your micro instance from:
613 MiB memory
Up to 2 EC2 Compute Units (for short periodic bursts)
EBS storage only
32-bit or 64-bit platform
I/O Performance: Low
EBS-Optimized Available: No
API name: t1.micro
to:
15 GiB memory
8 EC2 Compute Units (4 virtual cores with 2 EC2 Compute Units each)
1,690 GB instance storage
64-bit platform
I/O Performance: High
EBS-Optimized Available: 1000 Mbps
API name: m1.xlarge
The compiling time will be reduced to about 25 minutes.Once you got the instance backup with the m1.xlarge instance, continue like so:
Code:
sudo su
cd /usr/src/ubuntu*
fakeroot debian/rules binary
#Check if your deb files were created
cd ..
ls *.deb
#install new kernel
#IF A GRUB MENU POPS UP PICK PACKAGE VERSION
sudo dpkg -i linux-*.deb
Then shutdown your system again and change it back to a micro instance. Then boot it back up.
#Check your new Kernel version
Code:
uname -r
#Check if Kernel HZ value change persisted:
cat /boot/config-`uname -r` | grep HZ
#If value 1000HZ=yes then you did it right!
Step 4. Install Asterisk 11
Code:
#become root
sudo su
# Install dependencies:
apt-get install libiksemel-dev libsqlite3-dev libssl-dev libnewt-dev libxml2-dev
#get source
#note: dahdi needs to be installed to compile and install libpri -- we don't really need it for any other reason
cd /usr/src/
wget http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete/dahdi-linux-complete-current.tar.gz
wget http://downloads.asterisk.org/pub/telephony/libpri/libpri-1.4-current.tar.gz
wget http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-11-current.tar.gz
#extract source
tar zxvf dahdi-*
tar zxvf libpri-*
tar zxvf asterisk-11*
#resolve error for compiling dahdi
ln -nsf /usr/src/linux-headers-`uname -r`/include/asm-x86 /usr/src/linux-headers-`uname -r`/include/asm
#install dahdi
cd /usr/src/dahdi*
make && make install && make config
#install libpri
cd /usr/src/libpri-1.4*
make && make install
#install asterisk
#note: once the menu pops up check and make sure you have chan_motif and xmpp (should have a * next to them)
cd /usr/src/asterisk*
./configure && make menuselect && make && make install && make config && make samples
Step 5. Configure Google Voice
Backup original conf files (you should still be root)
Code:
cd /etc/asterisk
cp extensions.conf extensions.conf.orig
cp motif.conf motif.conf.orig
cp sip.conf sip.conf.orig
cp xmpp.conf xmpp.conf.orig
New Config files compiled by jhax01 - GO TO POST #85 FOR MORE DETAILS. Now you will want to replace the following files with these (change USERNAME to whatever you want and make sure you google account info is correct):
#extensions.conf - Don't forget the USERNAME on the last line
Code:
[general]
autofallthrough=yes
; If an unauthenticated request some how gets through, send them to free 411.
[default]
exten => 411,1,Answer()
same => n,Dial(Motif/google/[email protected])
[local]
exten => _1XXXXXXXXXX,1,Dial(Motif/google/${EXTEN}@voice.google.com,,r)
exten => _XXXXXXXXXX,1,Dial(Motif/google/${EXTEN}@voice.google.com,,r)
exten => _+1XXXXXXXXXX,1,Dial(Motif/google/${EXTEN}@voice.google.com,,r)
[incoming-motif]
exten => s,1,NoOp()
same => n,Set(crazygooglecid=${CALLERID(name)})
same => n,Set(stripcrazysuffix=${CUT(crazygooglecid,@,1)})
same => n,Set(CALLERID(all)=${stripcrazysuffix})
same => n,Dial(SIP/USERNAME,20,D(:1))
#motif.conf
Code:
[google]
context=incoming-motif
disallow=all
allow=ulaw
connection=google
#sip.conf - Pay attention to externhost, secret, and USERNAME
Code:
[general]
allow=all
allowguest=no
nat=force_rport,comedia
tcpbindaddr=0.0.0.0
tcpenable=yes
externhost=ELASTICIP
localnet=10.0.0.0/8
[USERNAME]
type=peer
secret=PASSWORDYOUGENERATE
host=dynamic
context=local
transport=udp,tcp
#xmpp.conf
Code:
[general]
[google]
type=client
serverhost=talk.google.com
[email protected]
secret=GMAILPASSWORD
priority=100
port=5222
usetls=yes
usesasl=yes
status=available
statusmessage="VOIP"
timeout=5
Code:
#
# Stop/Start asterisk
#
sudo /etc/init.d/asterisk stop
sudo /etc/init.d/asterisk start
If everything went at planned your Asterisk Server with Google voice should be working, you can now login with your SIP client utilizing the extension username and password that you chose in sip.conf.
BONUS STEP. PPTPD VPN
install (make sure you are still root)
Code:
apt-get install pptpd
Now take the following code and copy it into a script and execute as root:
Code:
echo "localip 10.40.1.1" >> /etc/pptpd.conf
echo "remoteip 10.40.1.20-50" >> /etc/pptpd.conf
echo "ms-dns 8.8.8.8" >> /etc/ppp/options.pptpd
echo "ms-dns 8.8.4.4" >> /etc/ppp/options.pptpd
echo "ms-dns 172.16.0.23" >> /etc/ppp/options.pptpd
pass=`openssl rand 8 -base64`
if [ "$1" != "" ]
then pass=$1
fi
echo "VPN pptpd ${pass} *" >> /etc/ppp/chap-secrets
echo -e "VPN service is installed, your VPN username is \033[1mVPN\033[0m, VPN password is \033[1m${pass}\033[1m"
Done. Just a reminder, do not upgrade the system to Ubuntu 12.04 or you will lose the recompiled Kernel. If you update the kernel you will need to recompile...

Eh? Understood none of that. What even is Asterisk?

CrazyPeter said:
Eh? Understood none of that. What even is Asterisk?
Click to expand...
Click to collapse
https://www.asterisk.org/

1. What is the advantage of this method over centos 6.3 + PIAF on EC2?
2. It seems you are using UDP transport on sip port 5060. How's the battery life when you are on 3g?

acegolfer said:
1. What is the advantage of this method over centos 6.3 + PIAF on EC2?
2. It seems you are using UDP transport on sip port 5060. How's the battery life when you are on 3g?
Click to expand...
Click to collapse
1. No advantage really, other than less resource are used. This is just using Asterisk 11 -- does centos/piaf combo use Asterisk 11?
2. Yes, I do use UDP because my SIP client is not running all the time. I mostly only use it for outgoing calls. The above steps could be easily adjusted for TCP use.

Or you can download the obi app from play store and use that with Google voice to make WiFi calls.
Much easier to setup all you need is your login info.
Sent from my Nexus 4 using xda premium

errorcod3 said:
1. No advantage really, other than less resource are used. This is just using Asterisk 11 -- does centos/piaf combo use Asterisk 11?
2. Yes, I do use UDP because my SIP client is not running all the time. I mostly only use it for outgoing calls. The above steps could be easily adjusted for TCP use.
Click to expand...
Click to collapse
1. I just managed to install centos + PIAF on EC2. It was a major struggle. It's using asterisk 1.8 but can add gtalk trunk with GV motif.
2. When csip uses TCP transport to connect to PIAF in EC2, I continue to experience the same hangup issue. I guess it will happen to your setup as well if you use TCP.

kthejoker20 said:
Or you can download the obi app from play store and use that with Google voice to make WiFi calls.
Much easier to setup all you need is your login info.
Sent from my Nexus 4 using xda premium
Click to expand...
Click to collapse
No offense, but there is a reason this thread was made. It was spawned by the efforts of TWO other threads as an alternative solution to making VOIP work RELIABLY over 3G.
OP, you should add a blurb at the very beginning of the post that notes why this thread was made so we can curb responses like this in advance.

Is there any advantage to doing this as opposed to getting service from a reputable VOIP provider (voip.ms, callcentric, etc) and just connecting to their servers? Surely their connection/service is going to be pretty reliable and call rates are more than reasonable (like 0.01 per minute).
These providers support G.729 so you could connect with csipsimple and have pretty good quality over 3G I think.
Are you guys doing this to avoid paying 0.01 a minute or to try to get more reliable service? I don't understand...

cmaxwe said:
Is there any advantage to doing this as opposed to getting service from a reputable VOIP provider (voip.ms, callcentric, etc) and just connecting to their servers? Surely their connection/service is going to be pretty reliable and call rates are more than reasonable (like 0.01 per minute).
These providers support G.729 so you could connect with csipsimple and have pretty good quality over 3G I think.
Are you guys doing this to avoid paying 0.01 a minute or to try to get more reliable service? I don't understand...
Click to expand...
Click to collapse
Yeah, we REALLY need that blurb about this in the OP.
We are trying to avoid any very small cost by doing this, although that was AceGolfer's original intent. The problem is that, so far, will all the previous setups using VOIP providers + GV + SIP client, there is always one thing eluding us: RELIABILITY. Really, the problem seems to stem from the fact that a single provider does not offer everything needed to make this work reliably on 3G. This stuff works great on WiFi. Even call out USUALLY work flawlessly. But INCOMING calls have been missed, traveling calls (in car) can be sketchy, and even 1 bad connection in 10 is just plain lousy.
Frankly, I can deal with some of those problems. Mine, and I'm sure many other posters', wives and business partner/clients, cannot. This is just another method to try to figure out the best and most reliable way to get VOIP on our phones.

cmaxwe said:
Is there any advantage to doing this as opposed to getting service from a reputable VOIP provider (voip.ms, callcentric, etc) and just connecting to their servers? Surely their connection/service is going to be pretty reliable and call rates are more than reasonable (like 0.01 per minute).
These providers support G.729 so you could connect with csipsimple and have pretty good quality over 3G I think.
Are you guys doing this to avoid paying 0.01 a minute or to try to get more reliable service? I don't understand...
Click to expand...
Click to collapse
Here's what we (smartphone users over 3g) want: G729 (or iLBC) and TCP.
The 2 VoIP providers that you mentioned only offer UDP connection, which will kill smartphone batteries. I have heard that some people even use pbxes as the gateway for voip.ms so that they can use TCP connection.
Fortunately, there are several methods we have found that don't cost anything.

acegolfer said:
1. I just managed to install centos + PIAF on EC2. It was a major struggle. It's using asterisk 1.8 but can add gtalk trunk with GV motif.
2. When csip uses TCP transport to connect to PIAF in EC2, I continue to experience the same hangup issue. I guess it will happen to your setup as well if you use TCP.
Click to expand...
Click to collapse
What's the hang up issue that you are experiencing? I can switch my configuration over to TCP and see if I get the same issue.

bluespire said:
Yeah, we REALLY need that blurb about this in the OP.
We are trying to avoid any very small cost by doing this, although that was AceGolfer's original intent. The problem is that, so far, will all the previous setups using VOIP providers + GV + SIP client, there is always one thing eluding us: RELIABILITY. Really, the problem seems to stem from the fact that a single provider does not offer everything needed to make this work reliably on 3G. This stuff works great on WiFi. Even call out USUALLY work flawlessly. But INCOMING calls have been missed, traveling calls (in car) can be sketchy, and even 1 bad connection in 10 is just plain lousy.
Frankly, I can deal with some of those problems. Mine, and I'm sure many other posters', wives and business partner/clients, cannot. This is just another method to try to figure out the best and most reliable way to get VOIP on our phones.
Click to expand...
Click to collapse
I have been using this setup for about 6 months. It's incredibly reliable. I'm the only one using this server as opposed to more main stream options where hundreds or thousands of people are using the service. Plus, I can use it for other things such as a VPN server when I need it.

ADD TCP SUPPORT
On your security group settings for the Amazo EC2 open port 5060/tcp.
login and execute:
Code:
sudo ufw allow 5060/tcp
Edit sip.conf and add the following under the [general] section:
Code:
tcpbindaddr = 0.0.0.0
tcpenable = yes
Under your [USERNAME] in sip.conf add the following:
Code:
transport=udp,tcp
restart asterisk
ADDED TO OP.

It looks like you're only using G711u/a, and GSM for codecs. Have you tried installing G729? Or allowing G722?

errorcod3 said:
What's the hang up issue that you are experiencing? I can switch my configuration over to TCP and see if I get the same issue.
Click to expand...
Click to collapse
When Csipsimple is registered remotely (obviously is the case for EC2) using TCP, it takes 30 seconds to hangup after pressing the "end" call button. The voice gets transmitted during these 30 seconds.
OTOH, the call hangs up normally, if CSS uses UDP or I use a different client which costs to use G729.

osi13 said:
It looks like you're only using G711u/a, and GSM for codecs. Have you tried installing G729? Or allowing G722?
Click to expand...
Click to collapse
No, I haven't tried it. Asterisk 11 supports G722. If anyone is interested in using it just add an allow=g722 to the sip.conf file under [general].
Does Google Voice support G729?

acegolfer said:
When Csipsimple is registered remotely (obviously is the case for EC2) using TCP, it takes 30 seconds to hangup after pressing the "end" call button. The voice gets transmitted during these 30 seconds.
OTOH, the call hangs up normally, if CSS uses UDP or I use a different client which costs to use G729.
Click to expand...
Click to collapse
I just changed my configuration to TCP and I could not reproduce your problem. My calls terminate normally..

errorcod3 said:
No, I haven't tried it. Asterisk 11 supports G722. If anyone is interested in using it just add an allow=g722 to the sip.conf file under [general].
Does Google Voice support G729?
Click to expand...
Click to collapse
gtalk trunk only supports PCMU (g711 ulaw), which is used at PSTN. asterisk will transcode PCMU to other codecs such as G729.

errorcod3 said:
I just changed my configuration to TCP and I could not reproduce your problem. My calls terminate normally..
Click to expand...
Click to collapse
Forgot to mention, I'm also using CSS.

[Q] Problem with Global Proxy

Hello:
Until a week ago I was using the ProxyDroid to use the internet through wifi using my laptop with Virtual WiFi router without problems. I could use all applications using the global proxy that adjusts ProxyDroid, including 1mobile market apk.
However, after using other WiFi connections at a conference, which did not require proxy, my cel can't access the previous network through its old proxy. I fear that installing and uninstalling some applications, something changed some internal settings and I can't reset my cel connection.
Something remarkable, to activate ProxyDroid at work, I need to declare some bypass for some IP so that they can see. These pages show the alert if this, what makes me think that the ProxyDroid is working ok and changing the global proxy cel.
I use HTC Nexus One with CyanogenMod-7.2.0-passion
Please, help...:crying:
Thanks

About ProxyDroid and iptbles
I notice that ProxyDroid 2.7.0 not chage the iptables chains. I configure PD and enabled it, but when I use adb shell and then iptables -L just show this...
C:\Developers\ADT-Bundle\sdk\platform-tools>adb shell
# iptables
iptables
iptables v1.4.7: no command specified
Try `iptables -h' or 'iptables --help' for more information.
# iptables -L
iptables -L
Chain INPUT (policy ACCEPT)
target prot opt source destination
Chain FORWARD (policy ACCEPT)
target prot opt source destination
Chain OUTPUT (policy ACCEPT)
target prot opt source destination
#
So, it don't work
Help....... PLEASE
I use HTC Nexus One with MIUI 2.3.7 ES

[Q] Temporarily avoiding poor connections

Hi,
Is there any way to simply kill, or disable, this feature (if you ask me, a deliberate bug) on Sense 5, 4.2.2 based ROMs?
I'm not after a band-aid fix such as dumb apps that only remove dhcp files in hopes of a permanent fix. I want to just kill this feature. I only have wifi turned on at home (llama turns off wifi when I leave home/work). As far as my APs, I have tried channels 1 - 11 (US) and I've even made sure to use unused channels in my area. Nothing works, I still get the post title error message.
So XDA, is this possible?
Thanks

After a bit of research, I have found a suitable solution.
Depending on your region, set wifi channel accordingly. Since I am in the US, i added this to build.prop:
Code:
ro.product.locale.language=en
ro.product.locale.region=US
ro.wifi.channels=11
I chose channel 11 because it's a channel infrequently used in my area. Additionally, I used an OSX app called iSSIDer to see which channels were being used in my area.
Since doing this, I have yet to see the error. Also, I disabled 5GHz on my AP (had both 2.4 and 5 running simultaneously) and set 2.4GHZ to automatic.
From what I've seen, most ROMs will leave ro.wifi.channels option empty.
Anyways, hope this helps others.
oh, btw, you can verify your region and permitted channels here:
http://en.wikipedia.org/wiki/List_of_WLAN_channels
If you're in the US, do not set ro.wifi.channels to channel 14. It is the forbidden fruit. You can't use it.
----- [ EDIT ] -----
Two more discoveries that have helped out tremendously:
1) flash radio with newest radio available to you, for your network (or use newest international radio).
2) Add the following lines to build.prop:
Code:
#
# Enable LTE
#
ro.config.lte=1
telephony.lteOnGsmDevice=1
ro.telephony.default_network=9
ro.ril.def.preferred.network=9
** Since I an AT&T customer, in the Los Angeles region, I've LTE coverage. I added the following to build.prop, since using GE Phone, the option for LTE is gone.