Which phone do you suggest has best performance with Skype out and Skype In
also do the Skype mobile versions support skype SMS and does this work well.
Battery issues and performance with Skype running on each model?
I have read a few things about the GPS functionality on P3600.. is this true?
Skype: The 4350 (Herald) only has a 200MHz processor, so technically the P3600 ought to be better as Skype is processor intensive. Skype 2.2 doesn't work properly with 3600; not sure about 4350. Skype 2.1 works fine.
Of course the major difference is that the 4300 has a hard keyboard. For a faster processor, wait for the Tytn II
GPS: There are no problems with the P3600 GPS. Most reported problems are either due to people messing with the default settings, or not configuring the Application settings of the various GPS programs properly.
Skype 2.2 works great with mine P3600. For chat only
Are You Serious?
Using Skype for Chat only, is, well,.....
Looks like Skype/Google folks are planning something. It´s plain deadbrain to kill some devices out of your "latest" SW upgrade.
Don´t you think?.
I just have had to pass 2.1 to some surprised friend that just bought a new P3600.
Unbealivable.
skype 2.1 on 3600 / trinity
So then Skype 2.1 works with trinity for voice calls?
also anyone has any experience of making Skype SMS work with Skype for Mobile>?
Looking Glass said:
Using Skype for Chat only, is, well,.....
Looks like Skype/Google folks are planning something. It´s plain deadbrain to kill some devices out of your "latest" SW upgrade.
Don´t you think?.
I just have had to pass 2.1 to some surprised friend that just bought a new P3600.
Unbealivable.
Click to expand...
Click to collapse
I have replaced it with v2.1 and everything's just fine now. Voice calls are OK.
tapanp said:
I have read a few things about the GPS functionality on P3600.. is this true?
Click to expand...
Click to collapse
No - it's not at all true. You have not read a few things about the GPS functionality on P3600 at all.
If you could explain what you think you read - I could have a go at answering.
TO let you know - with recent (since March) ROMs, the GPS functionality is activated and works very well indeed.
Anyone who tells you otherwise is lying!
tapanp said:
So then Skype 2.1 works with trinity for voice calls?
also anyone has any experience of making Skype SMS work with Skype for Mobile>?
Click to expand...
Click to collapse
That's what I meant, 2.1 works OK with voice calls.
I haven't tested the SMS feature.
skype 100% works even for voice chat through network
Hi I use my htc tytn & htc 3600 both work excellent with skype even through my network provider (3g gprs) for voice chat & text both. I am 3 network & voip is free for me on my x-series add on so i installed fring from www.fring.com & do voice cahtting with my buddies online on any of these services: msn messenger, googletalk, skype, icq & sip via 3g/gprs (i can do it through wifi as well but as i have unlimited voip minutes & i don't have to depend on wifi network) so i am online 24 hrs & people can call me free & i can call online people for free & the quality is excellent as good as a normal phone either through network or wifi. I installed skype for Imate Jas Jam (when u have to select the make & model) on both of my phones). if u need any assistance my email is: [email protected]
khurram41 said:
Hi I use my htc tytn & htc 3600 both work excellent with skype even through my network provider (3g gprs) for voice chat & text both. I am 3 network & voip is free for me on my x-series add on so i installed fring from www.fring.com & do voice cahtting with my buddies online on any of these services: msn messenger, googletalk, skype, icq & sip via 3g/gprs (i can do it through wifi as well but as i have unlimited voip minutes & i don't have to depend on wifi network) so i am online 24 hrs & people can call me free & i can call online people for free & the quality is excellent as good as a normal phone either through network or wifi. I installed skype for Imate Jas Jam (when u have to select the make & model) on both of my phones). if u need any assistance my email is: [email protected]
Click to expand...
Click to collapse
Lucky man!
Where in the world you get "unlimited" voip 3G access for free? (monthly flat rate I suppose) I want that too! ;o)
I'm sure you have spoiled lot of jealous readings on your post......
BTW do you have a reading on how much 3G Mb data your Voip calls use per minute?
Has anyone been successful in setting up the native SIP client on a tab running gingerbread??
I am looking for someones experiences with this native VoIP client and also can you tell me what SIP Service Provider you used?
This link describes how to configure the Android native SIP client on gingerbread devices. Perhaps someone would be kind enough to explore these settings and post your results.
http://www.voipvoip.com/android/sip.html
UPDATE: I just learned that the 'settings' menu for internet calls does not appear by default in gingerbread. Its been disabled on our SGTs. To enable you must copy a file to system/etc/permissions. Can someone here shed some light on this? Is this only true for the stock Gingerbread thru Kies download? thanks.
da_in_sd said:
Has anyone been successful in setting up the native SIP client on a tab running gingerbread??
UPDATE: I just learned that the 'settings' menu for internet calls does not appear by default in gingerbread. Its been disabled on our SGTs. To enable you must copy a file to system/etc/permissions. Can someone here shed some light on this? Is this only true for the stock Gingerbread thru Kies download? thanks.
Click to expand...
Click to collapse
I use Overcome ROM RC1 with enabled VoIP in Settings->Internet calls.
My provider is iptel.org.
I have an issue with incoming calls over SIP. I can't answer. I think the problem is in original dialer by samsung.
I'm using the Overcome ROM RC2 where the SIP option is enabled.
It works well, although the lack of keypad during a call makes it hard when you run into an automated phone service and you can't enter any dtmf tones. I would guess this is related to the problem cromed has when trying to answer a call. (I haven't received a call so can't answer first hand)
My only other grumble would be the low call volume, but this is also the case when using fring or sipdroid.
Audio quality though, is great.
Hmm. Has anyone tried one of the SIP clients on the market on their tab? Such as 'sipdroid' or CsipSimple ? Can you post your experiences here please? thanks
SIP Clients
da_in_sd said:
Hmm. Has anyone tried one of the SIP clients on the market on their tab? Such as 'sipdroid' or CsipSimple ? Can you post your experiences here please? thanks
Click to expand...
Click to collapse
I have used both Sipdroid and CsipSimple. Sipdroid is pretty basic but works as described. CsipSimple ate up my battery, and Sipdroid did until I switched to a PBXES account (uses TCP instead of UDP). PBXES is a pain to setup, but once you finally figure out how it works its fine. Used it to take my DX overseas a couple times with WiFi only (used a MiFi through XCOM Global), and have perfect voice. Need to use the Speex codec for 3G connections or else you get jitter or big delays; I use it for all connections though.
Just got the GB update on the DX and sad to see no native SIP client.
I now have a Polycom connected to PBXES for my home phone. Both are trunked through Sipgate.
da_in_sd said:
Has anyone been successful in setting up the native SIP client on a tab running gingerbread??
...
UPDATE: I just learned that the 'settings' menu for internet calls does not appear by default in gingerbread. Its been disabled on our SGTs. To enable you must copy a file to system/etc/permissions. Can someone here shed some light on this? Is this only true for the stock Gingerbread thru Kies download? thanks.
Click to expand...
Click to collapse
Well, I just upgrade SGT-P1000N to stock Gingerbread 2.3.3, and VoIP does not come enabled. Found the permission file at: http://forum.xda-developers.com/showthread.php?t=1016583
UPDATE: Yep, it works. Just tested it. Works with Bluetooth headphones, yay!
It may prove my naivete to even suggest this, but I have wondered if it might be possible to sign in to a Skype account with this native dialer. Fields are limited to username, password and server. So it's that server address that eludes me.
CostaRicaHobo said:
It may prove my naivete to even suggest this, but I have wondered if it might be possible to sign in to a Skype account with this native dialer. Fields are limited to username, password and server. So it's that server address that eludes me.
Click to expand...
Click to collapse
Nope, Skype uses proprietary protocol, not SIP.
Sent from my SGT-P1000N - XDA App - Gingerbread!
I'm having the same issues as mentioned above on Overcome Gold. No way to pick up an incoming call. Outgoing call is fine but no keypad if needed.
Quality is very good.
Have tried Sipdroid, Csimple,...and other...but my prefered has been Linphone. Excellent quality.
But now I'm using GrooveIp. Work well with Google voice and allows you to make free phone call coming and outgoing.
It's a paid app..but cost $2.00 on Amazon...$4 on Google Market. YOu may have to adjust some settings to fine tune it...I had to do that.
UPDATE: This may not work any more because Google said that they will cut off the Google Talk XMPP on 5/15/2014, which PBXES relies on.
This guide is not about how to make free calls. It's about how to make free calls using iLBC codec so that you can make reliable calls over 3g connection. If you only want the instructions, then skip to part V.
(edited) I have changed my VoIP set up to PIAF on AWS EC2. See my signature.
I. BACKGROUND
There are several known methods to make free VoIP (internet) calls over 3g. I'll briefly discuss each to explain why there's no perfect solution. I'm not claiming my solution is perfect, either. You have to try them out and find out yourself what works for you.
1. Groove IP app
codec: PCMU (aka G711u, uncompressed signal used by PSTN and Google Voice, only good for fast/reliable wifi connection)
pros: simple setup, works well on wifi. If you use VoIP only on wifi, this is for you.
cons: echo. The voice quality is terrible on mobile data (3g) connection.
2. talkatone app
codec: speex for standard quality
pros: easy setup, good quality on 3g using speex .
cons: doesn't integrate well with android
3. Vonage mobile app
codec: G729(?) when making regular calls. iSAC when calling a Vonage member.
pros: G729 provides an excellent quality even on 3g.
cons: can't receive calls and doesn't integrate with Android. You need to input 10 digit number to make a call, unless the number is in your contacts.
4. Google Voice/pbxes/sipdroid (http://forum.xda-developers.com/showthread.php?t=1791957)
codec: lots of options based on connection type
pros: excellent battery life (http://code.google.com/p/sipdroid/wiki/NewStandbyTechnique), If you choose speex codec for 3g, then call quality is pretty good on 3g.
cons: must unlock the screen before answering the call. doesn't support better codecs such as iLBC/G729.
5. simonics GV gateway https://simonics.com/gvgw/ + sip client of your choice
codec: PCMU
pros: simpler setup than pbxes, some ppl claim it's more reliable.
cons: limited codec support. no multiple registrations
6. Setting up your own asterisk server (http://enjoytechnology.info/)
pros: everything is configurable
cons: requires a 24/7 running server
I have been primarily using #4 setup but wasn't satisfied with the call quality on 3g so I was about to do #6. But I found a way to tweak #4 to use iLBC codec and greatly improved the call quality on 3g.
II. CODEC COMPARISONS
The call quality basically depends on (1) the codec that your phone uses and (2) the network throughput. Unfortunately, there's no perfect solution because of trade-offs.
PCMU (64kbps) is the uncompressed signal that is transmitted at PSTN. So it will give you the best quality as long as your network throughput supports it. This is the preferred codec for wifi connection. However, 3g network cannot consistently sustain 64kbps and will suffer from packet loss resulting in jitter. This explains why Groove IP works best on wifi but not over 3g.
At the other end of the spectrum is GSM codec (8kbps). It's designed in 1990 for wireless communication over 2g. It's the most compressed signal, which is used by your regular GSM providers (AT&T, Tmo). These wireless providers allocate sufficient capacity on voice calls so your voice calls will suffer little jitter. However, the call quality is inferior to PCMU codec.
There are many codecs in between. For example, Sipdroid and talktatone can use speex, which requires low bandwidth. These apps can send the same signal twice to compensate for packet loss.
The 2 codecs that haven't been used widely in commercial apps are G729 and iLBC. These 2 codecs require low bandwidth (<15kbps) but have algorithm to interpolate loss packets. It is suggested these 2 codecs are ideal codec for 3g connection. Here are some readings comparing G729 vs iLBC.
http://blogs.elastix.org/en/2009/11/...cs-in-elastix/
http://forums.whirlpool.net.au/archive/1413041
In short, both are excellent codecs for 3g. iLBC has a better quality but requires more CPU power, which Nexus 4 can easily handle. My setup relies on iLBC.
III. SETUP
My setup is an extension to #4 method above. It involves 4 free resources.
1. Google Voice (same phone number for in/out calls and free outbound calls using gtalk)
2. callcentric (VoIP provider for free inbound calls)
3. pbxes (PBX that manages in/out calls)
4. csipsimple (android VOIP app that supports iLBC)
Once set up properly, here's how the incoming and outgoing calls will work.
1. Incoming GV calls
Someone calls your GV number. GV forwards the call to callcentric. Pbxes intercepts the call and rings your phone. Your phone displays caller ID (CID) properly.
2. Outgoing GV calls
You place a call in your phone dialer. Pbxes calls the number using gtalk trunk. The recipient will see your GV number as CID.
IV. BENEFITS
What are the benefits of my setup?
1. Completely free in/out calls. You don't even give out any CC information. All I'm paying is 30/m for Tmobile's 100 min, unlimited data prepaid plan.
2. Excellent call quality even on 3g using iLBC codec. I often get better call quality than Tmo's GSM.
3. Keeping the same number for in/out calls.
4. Full Android integration.
5. Battery life. Only requires 1 SIP registration to pbxes using TCP port. Battery stat screenshots after 10 hours are attached here: http://forum.xda-developers.com/showpost.php?p=35889479&postcount=91
V. INSTRUCTIONS
Are you still with me? Then follow these steps.
#1. Get a fast phone that can handle iLBC codec. FYI, a 2-yr old Galaxy S has the sufficient CPU power for iLBC.
#2. Get an unlimited fast data plan such as Tmobile's $30 prepaid plan. http://prepaid-phones.t-mobile.com/monthly-4g-plans
#3. Install and set up Google Voice app on your phone. Without GV, you can't open a free pbxes account in #4.
#4. Install Sipdroid to create a pbxes account.
When Sipdroid opens, it will detect whether you have GV app installed. If so, it will give you an option to open a free pbxes account linked with Google Voice. Create the pbxes account. In case Gmail flags for an attempted login, allow pbxes to gain access. Once you create the pbxes account, make test calls in and out using Sipdroid. Some ISPs may block you from using VoIP so make sure you can make calls before the next step. As of now, sipdroid doesn't support iLBC codec so we will be using other sip clients for making calls.
#5. Log in to pbxes.org and click "personal data".
Fill out all the blanks. Doesn't need to be correct. If you skip this step, pbxes has been known to remove the account later. While you are doing it, I recommend changing the pbxes login password that you used in step #4 to something else. In addition, pick a data center (server) closest to you to lower latency. If you have registration issues, change to a different server. FYI, I had registration issues with www7 (Miami) server.
#6. Open a free phone number (DID) account at callcentric.com.
You can only get a NY area code number but it doesn't matter because you will be giving out your GV number to your friends. All incoming calls to callcentric (including GV forwarded calls) will be free. You can decline E911 service to stay completely free. Callcentric supports efficient codecs such as G.729 and iLBC. We will be relying on iLBC. The outgoing calls are not free but we will be using GV to make free outbound calls.
Next, we have to link pbxes with callcentric so that pbxes can receive GV forwarded calls to callcentric. There are two ways to do this. (If you are wondering why we are using callcentric instead of gtalk, it's because pbxes has issues with iLBC for gtalk incoming calls. Callcentric handles iLBC better so we are piggybacking it.)
#7a. Callcentric call forwarding to pbxes.
Log in to callcentric.com. Go to preferences>DID forwarding. Input your [email protected] in the box and save. AFAIK, callcentric doesn't charge call forwarding to a SIP address (our method). However, it does to a regular phone number.
Alternatively,
#7b. Have pbxes intercept the calls to callcentric.
Log in to pbxes.org. Click add trunk, select SIP trunk and input callcentric credentials. (user name starts with your callcentric 1777 number.) Select "audio bypass" to pass-through callcentric's iLBC signal to the phone. See the attachment for what my setup looks like.
Here's a tutorial on adding trunks: http://www4.pbxes.com/wiki/index.php/Getting_Started
#8. Add extension(s) in pbxes.
Log in pbxes.org. Click extensions. If you followed step #4, you should see extension "Sipdroid-200." You can add new ones or modify extension 200. Change the password to something that you can easily remember. This will be the password you will use to register your phone to pbxes in the next step. Finally, select "audio bypass."
#9. Install nightly version csipsimple app from http://nightlies.csipsimple.com/trunk/. The play store version crashed with iLBC.
First, add the pbxes account. If you are new to VoIP, use the wizard for pbxes. User name should be yourgoogleID-200, where 200 is the extension you used in #8. The password is what you changed to in #8. (I recommended you change the password in #8 so that you don't use your gmail password everywhere.)
Do not add callcentric account because having 2 registrations drains battery faster. If you followed the steps above, you will still be able to receive callcentric calls from your pbxes account.
I prefer csipsimple because it is has lots of options to configure. But it can be too overwhelming to some people. If you can't get it to work, then try other sip clients such as media5-fone.
#10. Let's change csipsimple for better call quality.
First, in settings/media/codecs, select iLBC as the only codec for 3g connection. If you have a strong wifi connection, then select PCMU for wifi connection.
(optional) If you experience looping when making calls, set up filters appropriately. I set up mine such that all calls will be made by pbxes except for 911.
#11. Google Voice (from webpage)
Add callcentric DID number (not the 1777 number). All the GV forwarded calls to callcentric will be forwarded to pbxes and ring csipsimple on your phone.
Uncheck Google chat so that pbxes only receives callcentric forwarded calls. FYI, GV doesn't support iLBC codec natively.
(optional) uncheck Tmo mobile number, if you are completely satisfied with this setup. If GV is forwarding to your cell, then your cell will be ringing twice and you may accidentally answer the cell phone using your minutes.
#12. (optional) Test in/out calls.
I use these numbers to test the audio quality.
*43 (pbxes echo test)
(909) 390-0003 (PSTN echo test)
(408) 647-4636 (record/playback)
Note that during a call in csipsimple, you can click the settings button in the lower right corner to change the in/out volumes and enable/disable echo cancellation.
You should also test incoming call quality using Google Voice call back. This is where I had the most troubleshooting. My incoming call quality was awful until I found callcentric trick.
I have tested this setup in many different scenarios including in a car running at 45 mph and very happy with the results. Of course, YMMV depending on how good your 3g connection is.
VI. LIMITATIONS
What are the drawbacks?
1. latency (audio delay). iLBC codec makes CPU interpolate to make up for loss packets. This is why the call quality is better than other codecs over 3g. However, the disadvantage is it increases audio lag. I'm experiencing a 0.5s lag one way on 3g. This is not too bad considering a cell>GV>cell call has a similar latency (http://www.howardforums.com/showthread.php/1621673-Voice-Latency-Test-Results, couldn't find a more recent test result)
2. Once a caller calls your GV number, it takes 5-6 seconds for your phone to ring. This is a limitation of PSTN and GV forwarding not VOIP. On the other hand, if a caller calls my cell number directly, it rings in 4-5 seconds. So all the call routing through caller>Google Voice>callcentric>pbxes>phone will add 1 extra second for your phone to ring compared with caller>phone.
3. Pbxes with gtalk trunk will make your Google chat status show as "online and available" all the time. Your friends may think you are online when you are actually not. Another side effect is you won't be able to answer GV calls from PC gmail any more. I haven't found any workaround to fix this other than creating a new google voice account.
4. Nexus 4 VoIP issues. It has been reported that several hardware mic functions (echo/noise cancellation) are disabled for N4 VoIP. http://code.google.com/p/android/issues/detail?id=41626 OTOH, echo cancellation is enabled in Galaxy Nexus. You can mitigate the issue with adjusting the media settings in sip client. But if you are in a noisy area, the mic is going to pick up all the noise. Consider using the headset.
VII. TIPS/TRICKS
1. If your family/friends know how to internet call, then it's best to give out your SIP address; [email protected]. Then the call will go out fast and the phone quality will be the best because you are not using PSTN at all. If both caller and recipient are using the same pbxes server, then pbxes is the only node between the two. For example, when my wife with the same setup calls me @pbxes, my phone rings in less than 2 seconds and the call quality is the best.
2. Create multiple extensions (300, 400, etc) in pbxes.
I have 3 devices registered to pbxes simultaneously: my cell phone, my tablet and the ATA for my home POTS phone. When I receive a GV call, all 3 devices ring simultaneously. And I can call any other extension independently by entering its 3 digit extension number.
3. Try other sip clients
I suggested csipsimple because it's open source and free despite being an unstable nightly version. You should try out other apps because they may perform better for you. For example, when I use media5-fone (free w/ ads, $5 to remove ads), the audio latency improved but it consumed a little more battery.
4. Battery improvement
My Nexus 4 easily lasts 16 hours w/ 3+ hour screen time. There are several things you can do to improve battery life. Use TCP only in client to communicate with pbxes.org. If you use csipsimple, you need to change the account settings. First, use the wizard to switch to expert mode. Then you will see "transport" option in the account settings. Select TCP. Another thing you can do to improve battery life is to experiment with larger keep alive intervals.
5. Use a different gmail account for your GV number
If you are uncomfortable with giving out your gmail password (even the app specific one) to pbxes, then transfer your GV number to another gmail account. http://support.google.com/voice/bin/static.py?hl=en&ts=1378507&page=ts.cs The whole process took less than 30 minutes. Once done, you don't need to create a new pbxes account. Just log in pbxes.org, go to Trunk>Gtalk and make the necessary changes. If done properly, you can make calls out again instantly. If you are using GV app, then you need to add the new account.
6. Delayed GV call forwarding
You can set up pbxes such that it rings your cell phone if you don't answer the internet call in X seconds. See
http://forum.xda-developers.com/showpost.php?p=36041116&postcount=196
VIII. Troubleshooting
1. This guide is definitely not simple. Lots of things can go wrong. If you want help, then provide as much information as possible. Posting a screenshot of pbxes call monitor log will be a good start.
2. If csipsimple doesn't work, try other sip clients such as media5-fone. If you can't make a call with media5 either, then you know at least it's not csip's fault.
3. If you suspect pbxes malfunctioning, change the datacenter (server) from pbxes.org.
Since you've done so much testing, maybe you could include some more information on how your setup works. Like, how many seconds does it ring? How about voicemail? Any issues if you enter/leave your home (i.e., wifi area) during a call, etc.
How about audio delay? Mine is noticeable using Google voice and the carrier.
Sent from my Nexus 4 using Tapatalk 2
I would like to test this setup. Can you give more details and examples for steps 3-b and 3-c in creating the trunk and inbound routing on pbxes.org?
Thanks again for any help that you can provide!
plee3
I'm very untested in this as well. Why not just port my number to the $30/mn plan. Then use gv from my PC to make calls or from groveip on my N4?
Sent from my Nexus 7 using xda premium
plee3 said:
I would like to test this setup. Can you give more details and examples for steps 3-b and 3-c in creating the trunk and inbound routing on pbxes.org?
Thanks again for any help that you can provide!
plee3
Click to expand...
Click to collapse
If you have never changed settings in pbxes, then this guide may be helpful.
http://lucas719.info/function/free_phone
If I get a chance in the future, I might update the OP with more instructions for those who have never used pbxes before.
marty331 said:
I'm very untested in this as well. Why not just port my number to the $30/mn plan. Then use gv from my PC to make calls or from groveip on my N4?
Sent from my Nexus 7 using xda premium
Click to expand...
Click to collapse
Because grooveIP uses PCMU and PCMA codecs, which won't work reliably on 3g mobile data. The biggest advantage of my setup is iLBC codec, arguably the best codecs for 3g mobile connection. Another great codec for 3g is G729 but I haven't found an integrated way to use G729 for free.
frigidazzi said:
How about audio delay? Mine is noticeable using Google voice and the carrier.
Sent from my Nexus 4 using Tapatalk 2
Click to expand...
Click to collapse
Latency, or delay, is inherent to all VOIP calls. To mitigate the delay, pick a server closest to you in settings from pbxes website.
post-mortem said:
Since you've done so much testing, maybe you could include some more information on how your setup works. Like, how many seconds does it ring? How about voicemail? Any issues if you enter/leave your home (i.e., wifi area) during a call, etc.
Click to expand...
Click to collapse
GV calls forwarded to pbxes ring about 17 seconds. This is well documented in another XDA thread.
I rely solely on GV voicemail.
When you leave wifi, you lose registration to SIP server.
I passed 2 tests today.
1. I can have a pretty good conversation while driving at 45 mph. I haven't had a chance to test on highways yet.
2. My wife is satisfied with the set up on her N4.
In addition, I found media5-fone app has less latency than csipsimple. However, you have to pay to remove the ads and it consumes more battery even w/ TCP connection.
acegolfer said:
I passed 2 tests today.
1. I can have a pretty good conversation while driving at 45 mph. I haven't had a chance to test on highways yet.
2. My wife is satisfied with the set up on her N4.
In addition, I found media5-fone app has less latency than csipsimple. However, you have to pay to remove the ads and it consumes more battery even w/ TCP connection.
Click to expand...
Click to collapse
Good news. Have you tried Bluetooth with media5-fone? I am having some issues using Bluetooth with csipsimple and am looking for another SIP client other than sipdroid.
Thanks... plee3
acegolfer said:
When you leave wifi, you lose registration to SIP server.
Click to expand...
Click to collapse
What exactly does that mean?
andoird213 said:
What exactly does that mean?
Click to expand...
Click to collapse
This means if you are on a call using wifi, when you leave your wifi range, your call will be dropped as your SIP connection is lost.
Hope this helps... plee3
plee3 said:
Good news. Have you tried Bluetooth with media5-fone? I am having some issues using Bluetooth with csipsimple and am looking for another SIP client other than sipdroid.
Thanks... plee3
Click to expand...
Click to collapse
I have not tried Bluetooth with N4. In fact, I don't know where my BT headsets are.
FYI, media5 is a free app with ads. It costs $5 to remove ads.
plee3 said:
I would like to test this setup. Can you give more details and examples for steps 3-b and 3-c in creating the trunk and inbound routing on pbxes.org? plee3
Click to expand...
Click to collapse
Seconded!
Thanks for sharing all these details, if I knew more about the pbxes setup, I might give this a shot. Any chance you'd share screenshots of your pbxes setup? I've tried modifying mine before, but never successfully.
quarksurfer said:
Seconded!
Thanks for sharing all these details, if I knew more about the pbxes setup, I might give this a shot. Any chance you'd share screenshots of your pbxes setup? I've tried modifying mine before, but never successfully.
Click to expand...
Click to collapse
3-b. Add a trunk using callcentric SIP credentials in pbxes. Log in to pbxes.org, click add trunk, select SIP trunk and input callcentric credentials. (user name starts with 1777 number.)
http://www4.pbxes.com/wiki/index.php/Getting_Started
3-c. Add an inbound route for callcentric trunk. This will pull any calls to callcentric and route them to N4. (OTOH, if you try to push the call from callcentric end, it will not be free.)
3-d. Select "audio bypass" in both callcentric trunk and extensions. pbxes will pass-through the calls from callcentric to N4 so that you can use iLBC codec. (pbxes claims to support iLBC but when I answer gtalk calls, there's no sound. That's why we need callcentric for incoming calls.)
This is what my pbxes looks like for 3-b and 3-c.
acegolfer said:
I have not tried Bluetooth with N4. In fact, I don't know where my BT headsets are.
FYI, media5 is a free app with ads. It costs $5 to remove ads.
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Thanks for the info, but the free version does not support Bluetooth (you need to upgrade to the paid version to enable Bluetooth).
Thanks again... plee3
How about vonage? free calls to numbers in the US
Just wanted to say I followed your guide and it appears to be working great.
Only issue I had was pbxe was sending calls to Google voice and call centric and my T-Mobile number. So I had a bunch of voice mails lol.
Think its all figured out now though. Incoming I have just call centric and out going to Google talk.
*edit
Just had a question, sure I am just being paranoid. But how secure is this setup? Like on the phone giving my social etc can someone listen in?
Also I am getting voice mail notificaion all of a sudden. Can't clear it and Google voice / tmobile show no new mail. Any idea?
Sent from my Nexus 4 using xda app-developers app
plee3 said:
Thanks for the info, but the free version does not support Bluetooth (you need to upgrade to the paid version to enable Bluetooth).
Thanks again... plee3
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Click to collapse
Ic. I didn't pay $5 to get the full media5 version. The little latency in csipsimple didn't bother me in real life settings.
intekmdma said:
How about vonage? free calls to numbers in the US
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Click to collapse
I heard good things about vonage app so I tested vonage on 3g a few days ago. The call quality of vonage was similar to my setup but it had more latency. I believe vonage is using either G729 or iLBC codec but the server is farther than my pbxes server. YMMV.
However, I found a bigger problem with vonage. I couldn't get it integrated with android. This means you need to open the vonage app and type in the 10 digit phone number to make a call. This won't work well when I'm on the fly, which is when I need vonage the most.
ogrillion said:
Just wanted to say I followed your guide and it appears to be working great.
Only issue I had was pbxe was sending calls to Google voice and call centric and my T-Mobile number. So I had a bunch of voice mails lol.
Think its all figured out now though. Incoming I have just call centric and out going to Google talk.
*edit
Just had a question, sure I am just being paranoid. But how secure is this setup? Like on the phone giving my social etc can someone listen in?
Also I am getting voice mail notificaion all of a sudden. Can't clear it and Google voice / tmobile show no new mail. Any idea?
Sent from my Nexus 4 using xda app-developers app
Click to expand...
Click to collapse
Not too sure about VOIP security. I'm relying on 3 services: GV/callcentric/pbxes so there could be a loose end. For example, I heard that Google "hears" everything. But any wireless company does the same.
I'm guessing you have a voice mail in pbxes because you said it's not GV or Tmo. To clear it, call pbxes voice mail by dialing *97.
In addition, I found csipsimple has a wrong voice mail input for pbxes. Here's how to change it, if you want to long press "1" to access pbxes voice mail or want to change it to GV voicemail number.
http://code.google.com/p/csipsimple/wiki/FAQ
In OP, I recommended pulling rather than pushing to avoid callcentric call forwarding charges. But I may have been wrong about it.
http://www.callcentric.com/features/unlimited_sip_uri_calling
If call forward to SIP URI is free, then callcentric can push the GV calls instead of pbxes pulling GV calls.
1. log in to callcentric.com
2. go to preferences>DID forwarding
3. locate the number that you added to Google Voice. click edit.
4. enter your pbxes SIP URI such as "[email protected]" and save.
I have made several test calls but don't see callcentric (which doesn't have my CC infor) is billing anything.
Why pushing (compared with pulling)?
1. It's faster. Once the phone call is placed, It takes 1 second less for your phone to ring. In my testing, the total time is now about 5 seconds, which is still slow because GV relies on PSTN. OTOH, if a phone call is made directly to @pbxes.org SIP, then it takes 2 seconds to ring. I have convinced my wife to call me using SIP URI.
2. It's more reliable. Pulling requires pbxes to register callcentric and intercept the calls.
3. It's simpler. You can skip steps 3-b and 3-c, which seem to be the confusing part.
I can't imagine they could bill you even if they wanted too without any kind of credit card info. I didn't even put a real name or address so I don't know.. lol.. But it does let me put in any number I want to forward too, I didn't actually try calling though because I am afraid of a bill. I imagine it would say something like this call can't be completed please add credit or whatever.
Its funny I followed your guide, doing all this reading.. for no real reason since i get like 3 calls a month.. Fun learning experience anyway.