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Log files on not making calls, Google Voice on Obi200

Started by VegasTech, June 16, 2018, 03:39:24 PM

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VegasTech

This being Saturday I had some time to study the Obi200/Google Voice problem.  I installed syslogd on my workstation and configured the Obi200 to log to my computer.

My basic setup is an Obi200 configured by obitalk.com for my Google Voice account.  I'm connected through a TM-AC1900 router to a ZTE Mobley device to the AT&T cell network.  I used the same Obi200 device for years through Centurylink without a problem, and I've had the problem of not being able to make or receive calls on and off since changing to AT&T.  I've tried bypassing the router, with no success.

Note that the obitalk.com dashboard says I'm connected to Google Voice, and the status in the Obi200 interface also says I'm connected, yet I still can't make or receive calls sometimes.

I'm logging with a verbosity of 7, so call details are somewhat lengthy.  Everything is identical through the dial sequence so I left that part off.  The only editing I did was to remove the timestamp and IP header information from each line, and to use XXXX for the last 4 digits of my phone number.  Here is the log output for a call that was able to go out.

CCTL:NewCallOn Term 10[0] ->1702625XXXX,1702625XXXX   
[DSP]: set FXS(0) DTMF detection level to (3)
[DSP]: set FXS(0) DTMF detection level to (3)
RTP:DtmfTxMtd:1(1),0
SIP/DLG:Start Early ICE on 18x
[DSP]: set FXS(0) DTMF detection level to (3)
[DSP]: set FXS(0) DTMF detection level to (3)
RTP:Start->4a7d273d:19305(80 0);0;0;0:0:0;0(46)
RTP:Set actpass 2
DTLS:Setup active
DTLS:Handshake Success
RTP:PeerRflxAddr=4a7d273d:19305
[SLIC]:Slic#0 ONHOOK

Here is the log for a bad call, which returned an error message recording saying that the service provider didn't respond.

CCTL:NewCallOn Term 10[0] ->1702625XXXX,1702625XXXX
[DSP]: set FXS(0) DTMF detection level to (3)
[DSP]: set FXS(0) DTMF detection level to (3)
PRD:1st down
[DSP]: set FXS(0) DTMF detection level to (3)
[DSP]: set FXS(0) DTMF detection level to (3)
RTP:Del Channel
[DSP]: set FXS(0) DTMF detection level to (3)
[CPT] --- FXS h/w tone generator (ringback)---
[DSP]: set FXS(0) DTMF detection level to (3)
[CPT] --- FXS s/w tone generator (sit_1) ---
[CPT] --- FXS s/w tone generator (sit_1) ---
[SLIC]:Slic#0 ONHOOK

Notice that the first 3 lines of both outputs are identical.  In the 4th line the good call returned:

RTP:DtmfTxMtd:1(1),0

While the bad call returned:

PRD:1st down

What are these logs telling me?

VegasTech

#1
I finally have a workable solution.

First, I was able to get a simonics.com account, so I connected to siminics on SP1.  Outgoing calls are 100% reliable but incoming calls are still hit & miss.

To fix the incoming call problem I forwarded incoming calls from my Google Voice number to a free callcentric number, then connected to callcentric on SP2.  That makes incoming calls 100% solid.

I didn't want any outgoing calls to go through callcentric because outgoing calls are charged a per-minute fee.  It's a low fee, but making calls to Google Voice via simonics is completely free.  To force all outgoing calls to SP1 I checked "Primary Line for Outgoing Calls" in the SP1 setup but not in the SP2 setup.  All outgoing calls seems to be routed through SP1 now.

This setup is somewhat Rube Goldberg, but will be fine for the time being.  I'll give it a month or so to see if conditions improve for direct connections to Google Voice and incoming calls from simonics.com.

One thing I've learned by all of this is that voip works better through some Internet providers than others, but I don't know why that might be.  Evidently there's a networking variable that we don't understand.

VegasTech

#2
I tried to connect to Google Voice directly with the Obi200 (not going through simonics.com) today.  Google Voice is working fine for both outgoing and incoming calls.

****Edited to Add****

Again, I spoke too soon.  It dropped again, requiring a reboot of the Obi200 to restore.  I'm back on with simonics.  Unfortunately Bill Simon informed subscribers today that he'll pull the plug in a week or so.

I'm experimenting with FreePBX/Asterisk right now.  Maybe I can get it to work.