So many of us have the Obi1xx series devices that recently stopped connecting to google servers due to a certificate update. This thread is intended to discuss the possibility of modifying the firmware to update the certificate and let these devices work with Google Voice again.
These devices are based on a MIPS-X processor similar to the Sipura ATAs and there is not a lot of tools/docs out there about them except for a Yahoo Group mostly related to DVD player chipsets. The venerable DogFace05 who was an expert with these types of devices once posted that he was able to extract this firmware sucessfully. Not sure if he is still around. Anyone else familiar with this architecture?
It seems that the place to start looking is the end of the firmware update file which contains some kind of table. Then there seems to be a loader section which presumably decompresses one or more other sections and loads them to RAM before executing the firmware.
So the questions are:
Can we extract, modify, and repack the firmware and create proper checksums/signatures?
Where is the certificate stored and in what format?
Can we drop in a new certificate without messing up other things (e.g. if the length of the certificate has changed) or do we need to move the certificate and patch the code pointing to it?
Is updating the certificate enough or is the codebase missing support that is necessary (e.g. if key length has changed)?
Anyone who wants to participate please post your thoughts.
Thanks
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ObiHAI Obi100/Obi110 Firmware Mod Discussion
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[PBX] FreePBX for the Raspberry Pi
The included script (install) and archive (install.tar.gz) will build
FreePBX 2.11, 12, 13, or 14 plus Asterisk 11, 12, 13, 14, or 15 on a
Raspberry Pi. iptables, dnsmasq, exim4, and pygooglevoice are also installed.
Installation takes a little over an hour to complete on a Raspberry Pi 3.
Download the latest Raspbian image.
For FreePBX 12, 13, or 14 Debian Stretch Lite is recommended:
https://downloads.raspberrypi.org/raspbian_lite_latest
For FreePBX 2.11, Debian Wheezy is required:
https://downloads.raspberrypi.org/raspbian/images/raspbian-2015-05-07/2015-05-05-raspbian-wheezy.zip
Write the image to an 8 GB or larger SD card. To accomplish this, I recommend Etcher or imageUSB:
https://etcher.io/ or http://osforensics.com/downloads/imageusb.zip
Create an empty file named ssh in the /boot/ directory (type NUL > ssh).
Connect the Raspberry Pi to your LAN using an Ethernet cable.
Insert the SD card and power up the Raspberry Pi.
Copy install and install.tar.gz to the /home/pi directory. To accomplish this, I recommend WinSCP:
https://winscp.net/eng/download.php
Using an SSH client, log in using pi:raspberry. To accomplish this, I recommend PuTTY:
http://www.chiark.greenend.org.uk/~sgtatham/putty/download.html
Make the install script executable:
$ chmod +x install
Run the install script:
$ sudo ./install
When prompted:
Set pi user password
Set root user password
Select FreePBX version
Select Asterisk version
Answer Edge option (FreePBX 13 or 14 only)
Answer IPv6 option ('No' recommended)
Review selections
Set Hostname (Item 2 / Hostname: FreePBX)
Set Localisation Options - Locale (Item 4 / I1)
Set Localisation Options - Timezone (Item 4 / I2 - in US, use America, not US)
Expand Filesystem (Item 7 / A1)
Finish / Reboot Now: No
The Raspberry Pi will reboot.
Log in as root.
If desired, enable PuTTY logging when prompted.
The system will be updated and then reboot.
Log in as root.
If desired, enable PuTTY logging when prompted.
Confirm install.
Installation will proceed unattended and then reboot.
Log in a root.
Installation will complete.
Utility scripts included in /root:
abn / dbn / ebn / ibn / qbn
===========================
Add / Delete / Export / Import / Query Blacklist Number
add-fcc-blacklist / del-fcc-blacklist
=====================================
Add / Delete FCC Blacklist
exclusions.fcc
==============
Numbers to Exclude from FCC Blacklist
ipt-add / ipt-del / ipt-chk / ipt-dsp
=====================================
Add / Delete / Check / Display iptables Entries
cell-phone-presence-bt / cell-phone-presence-obi
================================================
Cell Phone Presence Detection
pbx-backup / pbx-restore
========================
Backup / Restore PBX Configuration
image-backup / image-check / image-set-ptuuid / image-shrink / image-mount
==========================================================================
Backup / Check / Set PTUUID / Shrink / Mount an Image of the System SD Card
upgrade
=======
Upgrade / Update Linux
asterisk-13to14
===============
Upgrade Asterisk 13 to Asterisk 14
asterisk-upgrade
================
Upgrade Asterisk
set-boot
========
Set Boot PARTUUID (or /dev/mmcblk0)
set-timezone
============
Set System and PHP Time Zone
regen-ssh-keys
==============
Regenerate SSH Keys
clear-cache / clear-logs
========================
Clear Cache / Logs
install-fax
===========
Install Hylafax Server
add-fax-extension
=================
Add Hylafax Extension
del-fax-extension
=================
Delete Hylafax Extension
purge-fax
=========
Purge HylaFAX Server
HylaFAX fax server
==================
1. Execute install-fax: ./install-fax
2. Execute add-fax-extension: ./add-fax-extension
Multiple fax exntsions may be added to support simultanous sending and/or receiving of faxes.
SendFax
=======
SendFax is a program to send a fax file from Windows to a HylaFAX fax server.
No installation is required and no changes are made to your system.
Supported file tpyes are pdf, ps, tif, and tiff.
A cover page can be generated and prepended to outgoing faxes.
Leaving 'File to Send' empty will send only a cover page.
To configure, click Edit -> Options:
IP Address: (the IP address of your HylaFAX server)
Port Number: (the port number of your HylaFax server, normally 4559)
Username: (your username on your HylaFAX server, normally root)
Password: (your password on your HylaFAX server, normally blank)
Email Address: (the email address to deliver notifications to)
Notifications: (notification types to be sent)
Page Chop: (which pages to chop trailing whitespace from)
Threshold: (minimum trailing whitespace (in.) before chopping is used)
Modem: (which modem to use for outgoing faxes, normally blank)
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[General] Help with sending and printing faxes
Hello,
Here is my setup: a desktop, connected to a printer Brother MFC J5720DW. I use Microsoft Outlook 2016 pro to manage business contacts. Into each business contact there is its telephone number and fax number. For the telephone calls, there is an app integrated to Outlook. Inside Outlook I click to call the contact over VoIP (no phone line is necessary).
Is there any way to configure the Windows 10 Fax Utility to send faxes over VoIP without need of a phone line? Or is there any 3rd party app that runs on Windows and will do this job?
any advice is welcome :)
IM
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[Anveo] AnveoDirect vs. TelecomsXchange
Ward's post at http://nerdvittles.com/?p=24702 caught my eye.
It's in true Ward Mundy style (verbose and replete with plenty of overly-excited marketing-esque hype, but with more than a nugget of useful information :p), and my brain isn't doing well processing it right now, so I thought I'd toss it out here.
At first blush, it seems to offer basically the same kind of thing that AnveoDirect does (granular control over which routes are used, a very good LCR module, at-or-near-wholesale pricing, etc.), albeit at apparently even lower rates than AnveoDirect.
Is there a catch I'm missing? Or is this, as Ward always effectively says in just about every post about a new product on his blog, truly the Next Best Thing™?
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[Future9] Expired SSL Certificate
Good day, Nitzan.
The SSL certificate for your web site expired this past Saturday.
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[Equipment] Does anyone have experience with Cisco's SRP521W ATA?
I picked up a handful of these..
Reading everything, it appears that one can not dialplans for using the FXO port for outgoing calls.. If the proxy is offline or the power is out, only then will outgoing calls use the FXO port..
This is fine for what I want them for..
I want to use the ATA as any ATA gets used for.. Connecting a POTS phone, use a VoIP provider for all incoming and outgoing calls..
The building however is a MDU, the telephone jacks are connected to the door buzzer/intercom..
I am hoping that I can use the FXO port for the intercom system.. When it rings, it will ring through to the phone? Otherwise it doesn't get used.
Any configuration 'gotchas' in these units?
Will they accept the same HTTPS certs that the other Sipura/Linksys/Cisco VoIP devices do?
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[VOIPo.COM] Voipo intermittent issues making outgoing calls
I am having my first issue with Voipo with outgoing calls intermittently tonight. It has started working again for now. I have opened a support ticket with Voipo to see what they have to say. I have no problems at all receiving calls which is good. My Voipo provided Grandstream HT802 adapter is currently on the DMZ on my Sagemcom 5260 Fast Router.
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Hotel VOIP change over. Consultants? List?
I posted @ weeks ago re: @60 unit Hotel primed for a total VOIP change over from 60's copper system.
I've installed excellent Internet to units and plenty to building, staff, desk, etc.
Anywhere to look for experts that will consult on this? List somewhere?
Hotel Owners want to save money and I'd like to make money.
I'll be glad to elaborate further here and in detail via email
Thx
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Seeking opinions on VoIP Monitor (voipmonitor.org)
Can anyone give a review on this software?
I put a lot of work into setting up HOMER SIP Capture in my company's environment only to find it disappointing. It doesn't handle video sessions well, the UI is annoying, and the query engine is super slow no matter how many resources I throw at it. I'm ready to move on to something else.
VoIP Monitor seems to have more polish on the UI, a better capture agent, and a feature we really want, which is RTP analysis.
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what can i add to fix "transferring" calls?
i shouldnt use the word fix...more like change. using services such as callcentric and anveo, when a call comes in, we have to transfer the call to another extension.
what kind of unit could I add to our setup that the voip phones could connect to, then i could just put the line on hold and someone at another phone could just pick it up? TIA
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Allo Blox Open Source SBC
https://www.allo.com/sbc.html
http://www.blox.org
Looks interesting.
Anybody tried it?
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[CallCentric] activating/deactivating call treatment/ivr from phone
is there a way to pre-program turning on/off the call treatments or ivr from a phone instead of logging on to callcentric account to adjust?
would love to have a way for employee's to pick up phone and type in a code to turn off/on after hour call flow without giving them login access to the account.
obviously if we got it/left at same time everyday I would program it to set hours...however we come and go depending on if we got clients or not
TIA
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stubborn Polycom phones won't use failover route
Hi ,
We a PBX hosted on the internet (e.g., it's not on the local LAN) with a large number of a Polycom phones.
We have a failover route for when our primary internet goes down, which works for everything else ,including X-Lite softphones to the same Cloud VoIP provider.
However, the polycom phones are stubborn, and even with a reboot they want to use the old route somehow. Forcing them to get a new LAN IP address is the only way I can make them connect properly to the failover ISP.
The failover is transparent as far as the devices are concerned (e.g. the default gateway and DNS doesn't change). I'm using Ubiquity ER-X to manage the failover rule.
Sidenote 1 :
I believe the ER-X doesn't actively block devices from continuing to use the failover route the main comes back. That would be a nice feature to change if I could figure that one out.
Sidenote 2:
Another weird thing, if I run a continuous ping on my computer, those pings will "stay" on the old connection until I ctrl+c and start again ... I thought these pings were independant entities but maybe the source port changes or something like that.
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[Voip.ms] How to get CSR from Teleblend for Number Porting
Does anyone have advice for how to get the CSR (Customer Service Record) from Teleblend? I have contacted support but they don't reply. I tried calling, but the voice mail says contact support.
I have been trying to port my numbers for over one month. The current rejection is "billing telephone number", even though I put in the correct number. The CSR will show the correct billing #, that's why I need it.
I'm thinking I'll just start guessing numbers at this point because I don't know what else to do...
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Provider that offers whitelists
I am looking for a VoIP provider that has a whitelist feature. Currently I have Phone Power and an Obihai 200 ATA. I have been happy with the current system but we get alot of spam callers and it's driving me crazy. Our previous provider, Charter, had a whitelist but it was only for 10 numbers. I'd like more than 10 numbers. I have searched but didn't find much. I was hoping someone here had some advice.
Thanks.
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[Asterisk] Google contacts sync to mysql
Hi. New to the forum. I have an asterisk installation with Mysql and i would like to regurarly download my contacts from the Google to MySQL DB i maintain.
I have seen a post 'http://www.dslreports.com/forum/r30153045-PBX-Google-Contacts-to-MySQL-Database-Script-OAUTH2' and i tried to run the script but it returned a gazillion of errors.
Has anyone tried something similar, please?
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[Voip.ms] Yeastar S20 with voip.ms
Not having much luck getting trunks to register. Any help with how to do a trunk setup would be appreciated. Thanks in advance.
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[Unlock] New Grandstream ATA used by Vonage [Unlocked]
Looks like Vonage uses a new Grandstream ATA for their service, model HT802.
It's Grandstream newest generation of ATAs, with 2 ports.
Currently available at Walmart and BestBuy for $10.
If unlockable, this could be a nice, inexpensive 2 port ATA. Interesting that they are using a MicroUSB port for power.
I don't have a way of getting any of these soon to test as I am located in Canada. However, I am sure there are similarities between unlocking those and the BasicTalk HT701 (though probably they will not accept Mackey's firmware without some modifications).
Edit: seems like some new members are jumping into unlocking these adapters without going through the thread and reading all the comments. The proper unlocking method, tested by many already is posted here.
I still recommend reading through all the posts to learn from other members experience before starting anything.
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obi 100 sp1 and sp2 ring sounds
Is there a way to configure a different sound for when SP1 and SP2 ring?
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[Anveo] AnveoDirect vs. TelecomsXchange
Ward's post at http://nerdvittles.com/?p=24702 caught my eye.
It's in true Ward Mundy style (verbose and replete with plenty of overly-excited marketing-esque hype, but with more than a nugget of useful information :p), and my brain isn't doing well processing it right now, so I thought I'd toss it out here.
At first blush, it seems to offer basically the same kind of thing that AnveoDirect does (granular control over which routes are used, a very good LCR module, at-or-near-wholesale pricing, etc.), albeit at apparently even lower rates than AnveoDirect.
Is there a catch I'm missing? Or is this, as Ward always effectively says in just about every post about a new product on his blog, truly the Next Best Thing™?
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