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FreeIP Calling Help - IPKALL w/PBXes setup

raiderleaf

Member
Hello All,

I've been trying to get my Google Voice unlimited calling set up on my Galaxy Note 2 and I'm having trouble with the IPKall and PBXes part of it.

Problem Statement:
When I add the IPKall number to Google Voice, it asks me to activate it (THIS IS NOT THE SAME ISSUE AS OTHERS HAVE HAD REGARDING THE NUMBER BEING USED ON MORE THAN ONE GV ACCOUNT). When I click to activate it, my android phone doesn't ring (doesn't matter if I use CSipSimple or Sipdroid, both of which are hooked up to PBXes).

**As a note, I've tried other programs such as Talkatone, Spare Phone, and Groove IP and the call quality was completely garbage.

Here's what I've done:

1. Generated a 1-777-XXX-YYYY number for CallCentric
2. Took that number to IPKall to get a US-based phone number (206-XXX-YYYY)
3. Went to PBXes to set up the account (as a note, I have the legit GTalk Trunk for outbound calls)

PBXes Details:

Extensions:
Extension Name: SIP_CLIENT
Extension No: 200
Device Password: = PBXes Password
Username: = PBXes Username - 200 (same as extension no.)
DTMF Mode: Auto
Audio Bypass: No
Dial: SIP/PBXes Username - 200

Trunk Details: (For Inbound Calls)
Trunk Name: IPKall
dtmfmode: Auto
Audio Bypass: No
Username: 206XXXYYYY (IPKall Phone Number)
Password: ABCDEFG (IPKall Account Password)
SIP Server or Proxy: voiper2.ipkall.com
Domain: [Blank]
Register: Yes (inbound and outbound calls).

Inbound Routing:
Trunk: [BLANK]
Caller ID Number [BLANK]
Regular Hours:
Extension: SIP_CLIENT <200>​
Force Regular Hours (Selected)​

At this point, everything should work considering I'm having problems with receiving inbound calls. Once again, the issue is that PBXes isn't forwarding the IPKall call properly (my android isn't picking it up even though it's logged in to PBXes on the CSipSimple / Sipdroid client). Both clients do not run at the same time, rather I've alternated to see if it made a difference.

Please help if you can!

Thanks!

-raiderleaf

As a note, here are the references I've consulted:
How to Get a US Local Phone Number
How to setup PBXes, step by step | VoIP Forum
http://androidforums.com/android-applications/5673-sipdroid-set-up-help-2.html
 
I can't answer your question, but I suffered through all of that SipDroid, CSIPSimple, PBXes.org crap for a long time (and had SIPDroid working for a few months) before it crapped out and I switched to the free version of GrooveIP. I can't say that there was any difference in call quality between GrooveIP, CSIPSimple, NetTalk, and any other SIP app. That SIP-client code isn't rocket-science. I'd say that about 99% of the call quality is related to your Internet connection. Good luck with the torture session.
 
I can't answer your question, but I suffered through all of that SipDroid, CSIPSimple, PBXes.org crap for a long time (and had SIPDroid working for a few months) before it crapped out and I switched to the free version of GrooveIP. I can't say that there was any difference in call quality between GrooveIP, CSIPSimple, NetTalk, and any other SIP app. That SIP-client code isn't rocket-science. I'd say that about 99% of the call quality is related to your Internet connection. Good luck with the torture session.

When you say it crapped out, what do you mean?

I tried GrooveIP (premium version) and the call quality was so horrible that people told me that they wouldn't even talk to me anymore if I couldn't fix what was going on. I worked with the developer for over a month and no solution ever emerged. I sounded like total garbage to anyone I talked to.

The issue I'm having now is the following:

I have things working in the following manner: CallCentric --> PBXes --> CSipSimple/SIPDroid

The problem is that I'm still not receiving 100% of the calls to my device. It seems like PBXes routes only some of the calls. Additionally CSIPSimple sometimes doesn't try to reconnect to re-register the SIP and requires that I manually close it out and restart the app for it to re-register.

Tips anyone?
 
When you say it crapped out, what do you mean?
After spending great effort to get SIPDroid working with a free PBXes.org account, one day PBXes.org decided to change something (I can't remember what) and it broke. PBXes.org gave no warning or anything. They don't give a crap about their customers-- at least not their free customers. A bunch of people were complaining about it on the web. I worked hard to fix it but couldn't. I tried the free version of GrooveIP and never looked back. My call quality with Groove IP is VERY good, except for a bit of lag-- which I don't think is any worse than any of the other SIP solutions or even any worse than typical cellular delay.

Are you seeing a noticeable improvement in call-quality with solutions other than GrooveIP?

Again, I think that your Internet connection (speed and latency) and your WiFi router's settings (WMM) are the biggest barriers to call quality-- not the SIP client and PBX.
 
After spending great effort to get SIPDroid working with a free PBXes.org account, one day PBXes.org decided to change something (I can't remember what) and it broke. PBXes.org gave no warning or anything. They don't give a crap about their customers-- at least not their free customers. A bunch of people were complaining about it on the web. I worked hard to fix it but couldn't. I tried the free version of GrooveIP and never looked back. My call quality with Groove IP is VERY good, except for a bit of lag-- which I don't think is any worse than any of the other SIP solutions or even any worse than typical cellular delay.

Are you seeing a noticeable improvement in call-quality with solutions other than GrooveIP?

Again, I think that your Internet connection (speed and latency) and your WiFi router's settings (WMM) are the biggest barriers to call quality-- not the SIP client and PBX.

I'm sorry to hear that about SIPDroid and PBXes. I think what you're referring to is when they deleted the inbound GTalk trunks for free members. That definitely made things a pain in the ass when before all you needed was Google Talk and that's it (no DID's, forwarding, bla bla bla).

I have a Verizon tower literally within an eyeshot of my house, so my 4G coverage is absolutely pristine. Likewise, I have 18Mbs service from ATT Uverse (with a very high quality installation, differential wiring instead of single ended). Despite this, I had crap-tier call quality. I could hear people okay (and I mean only okay, not great) but people said I sounded like I was talking through a machine that was grinding. Same results over 4G or WIFI. I worked with SNR for over a month before I said there has to be a better way.

At any rate, I think I might have fixed my problem by enabling the STUN server and changing the TCP packet timing. I will have to wait and see if I get all of my calls tomorrow and then I"ll know if we have a winner. For what it's worth, when I use the Where's My Phone? service online to call my phone, the new config has had 0 failures out of 10 tries so far. I used to be around 20% failure, however it still is too early to tell.
 
For what it is worth - what I have read that GrooveIP's problem is that they use the default GTalk VOIP codec which is designed for an Ethernet connection and doesn't do well over WiFi or other higher latency connections.

I set up 2 PBXes accounts recently using SIPDroid and have inbound calls just using GVoice (although I replaced SIPDroid with CSIPSimple) but the performance of that has been disappointing. I think my next step will be to set up an Asterisk server and see how that works out.

I probably should just break down and buy an ObiHai device though.
 
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