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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Needing some help on call issue
Hello Guys
I do provide voip service over my wireless network and I have one customer is saying that when people calls into her phone nobody can hear her or calls drop out during middle of the call. I am using freepbx and cambium ata adapter.
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[Voip.ms] You can now port your iNums over to VoIP.ms...
All the team at VoIP.ms is excited to announce that we can now support iNums coming from other providers that have or will discontinue to support this great project.
While we are fully aware that the underlying carrier will stop offering support for the iNums initiative at some point in the future, we found value in building a porting process for it - a grande première in the VoIP industry!
Never heard of the iNum initiative? This project started back in 2008 with the objective to create a global dial code for IP communications (+883). iNums are free and can be used just like regular DIDs. Please note that there will be a one-time porting fee of USD$15.00 per number (hard cost from the underlying carrier) and that customers can only have one iNum per account. To kick start your iNum porting, please send us an e-mail at ports @ voip.ms.
For more information, you can read our Wiki article here or directly reach out to our support.
Thanks for being a VoIP.ms customer and we hope you will enjoy this announcement.
Best regards,
VoIP.ms Team
Latest News and Announcements
(There is no relevant Wiki article... yet! OE)
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[Equipment] Multiple lines on a phone (How?)
So I purchased a couple of Grandstream GXP-2140 phones a while back and am just starting to get them setup.
Using FusionPBX, mainly because I dabble with SignalWire and there Slack channel also has great freeswitch support, and I have gotten provision working with the phone. Still need to dive in deeper for some more features, but that will be done over time.
So my question is, what is the proper way to do multiple line availability on a phone like this. Is it assigning and registering multiple extensions and just labeling them "Line 1", "Line 2", "Line 3", etc.? Or a better way of doing it?
Thanks in advance for any input.
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TIP: Windows softphones: Windows strips DSCP by default
By default, Windows (XP and later, at least) strips TOS/DSCP (QoS) markings applied to IP packets by applications. You can work around this by following the instructions here:
http://www.iperfwindows.com/IPERF-QoS-tests2.html
Of course, change the App name from IPERF.
The one thing that I could not get to work (Windows 7 x64 SP1) was allowing the application's DSCP marking to override the QoS policy DSCP marking. I created two Windows QoS policies (X-Lite, Zoiper) that set DSCP to 46 (Expedited Forward, EF) for all UDP packets from the application. Works fine.
If anyone gets the application marking to override the QoS policy marking please let me know, including the version of Windows that you are running.
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Old VoIP companies: Dead or Alive ??
Things on the Internet never seem to die.
Just wondering what is going on with some of these old outfits, are they still in business or are they ghosts?
Ideasip
http://www.ideasip.com/
Site labelled 2006, most recent news posted 2010.
Teleblend (probably a few stragglers left from Sunrocket....)
Site labelled 2015.
http://www.teleblend.com/
Vestalink (aka Intelafone, Obivoice)
Site labelled 2019 but only shows a login page.
https://www.vestalink.com/login.php
And so forth.
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Which service to port/migrate off of Ooma?
We have had Ooma service now for 8+ years but as we use our home phone less and less are looking at migrating our home line to a new service.
I have been considering Google Voice but am confused about state of gateway support for Google and also don't want to spend a lot of time handling a service migration followed by lack of equipment support.
What currently does the community like for low cost residential support? In all cases will it be necessary to port out my home # to an intermediary like a pre-paid mobile SIM before migrating it to another service?
Currently with Ooma I am using the simulataneous ring feature to my mobile and the blacklist feature (which doesn't seem to do much). I also occasionally fax on Ooma.
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Dial plan adjust?
I have 3 Grandstream phones, I also use voip.ms.
I can call from one phone to another by using the extension I have assigned each phone. This works well other than there is a long delay between dialing the extension and the call going thru. eg ext 101
Is there a way to modify the dial plan to shorten this? Right now the phones just have the factory dial plans and other than that they seem to work fine.
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[Future9] DID released for non payment?
I have a DID through Cheapvoipinc.com. I'm very happy with the service.
My account has a +ve balance.
I received an email this morning from future-nine.com saying that the DID has been released for non-payment.
Anyone else?
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Virtual PBX Recommendation
VOIPo has a virtual PBX offering, but it's in BETA and cannot be relied upon. The monthly price is low, but the per-minute call pricing is average or higher. We need a virtual PBX with a main number (IVR) and four or more extensions. Can you recommend a low-cost provider?
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Voipo down: Two lines
Getting dialtone but no outbound or inbound service I know they are migrating datacenters but there is no outage information on their website. Online chat showed active but my chat request showed as in queue and then disconnected. The latter is unacceptable - if chat is not available, at least fess up.
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[Voip.ms] Guide Cisco 7941 getting started
Guide to getting VOIP.ms working on Cisco 7941
After buying my 7941 on ebay, I was driven mad trying to get this phone working on voip.ms. The guides on the internet were incomplete or using other services. I figured I would give my knowledge out there to all that want a cheap voip phone. (these go for ~$15 on ebay right now)
Step 1) When you get your phone, it most likely will have firmware on it for Cisco's call manager. This is not what we want. We want the SIP firmware to connect to voip.ms. Pay attention here because you cant use the latest firmware, the phone wont register with it. To get the firmware, go to: https://software.cisco.com/download/home/280083379/type/282074288/release/8.5(2)SR1
The firmware on that exact page 8.5(2)SR1 is what you will want. The exact file you want is the one that says "Phone Firmware Files Only - Compatible UCM" (the bottom one) Do not download the top one its not needed. In order to get this file, you will need to register an account with cisco. This account is free, but you must provide actual information as cisco's servers will look for falsafied info. If you get an email after you register saying your account must be verified, dont panic, i had that too and its a small waiting game for it to activate.
Step 2) Now that you have your firmware, You will need a TFTP server to upload it to the phone. This step is going to be cut into two parts. You will follow one or the other depending on your situation with your phone. If you can access the settings menu in your phone (settings button above the volume control) press it and then type * * # on the phone's keypad and go to Network Configuration >> IPv4 Configuration >> make sure DHCP is enabled and then scroll down to Alternate TFTP set that to yes. If you CAN'T set it, go to Step 2B. If you CAN set it, continue with step 2A below and skip 2B.
Step2A) This step assumes you can set "Alternate TFTP" to "Yes" under Network Config >> IPv4 Config. Scroll down to TFTP Server 1 (its in the same settings menu as the "Alternate TFTP") press edit and set the IP address you will be using for the TFTP server (a windows desktop computer is fine.) To enter the periods in the IP address, use the * button. Now on the desktop computer you secified in the phone, you will need a TFTP server. I recommend SolarWinds TFTP server. (https://www.solarwinds.com/free-tools/free-tftp-server) Once installed create a folder on your desktop named cisco and unzip the files from the zip we downloaded earlier into the folder. Put the XMLDefault.cnf.xml file i have for you in the folder with the files. Now in SolarWinds, hit file and configure and you should see start and stop buttons. Press stop if its started and at the botton fo the window hit browse and select your cisco folder on your desktop. Now hit start at the top and close the window. Now, on your phone press settings then * * # * *. The phone should reboot. It will boot normally and then reboot itself again. Don't press anything it should reboot into an upgrading screen and begin installing the firmware you downloaded.
Step2B) This step assumes you cannot set the TFTP server in the settings menu or you cannot access the settings menu at all for whatever reason. For you, there is a youtube video that explains how to get the firmware installed. https://www.youtube.com/watch?v=OTdeb1YMtsI Make you you use the firmware we downloaded in step 1!! After you follow these steps and get the proper firmware on the phone press the settings button then press * * # on the keypad and then go to Network Configuration >> IPv4 Configuration >> Alternate TFTP. Set that to yes then Settings button >> Network Configuration >> IPv4 Configuration >> TFTP Server1 and set this to an ip your will use for the tftp server (can be a windows desktop computer). (use * key as the decimal point)
Step 3) Alright! we got our firmware installed and we are almost done! Now time to get this puppy configured. I have included an XML file named "SEPMAC.cnf.xml" Go to settings on your phone and then network configuration. Your phone's MAC address should be shown as the 3rd setting down. in the downloaded XML file (SEPMAC.cnf.xml) replace the MAC portion of the filename with this MAC address. Example: SEP00670B84A05.cnf.xml. Next, right click the file and select edit (or open it in notepad or your favorite ACII editor). You'll wanna use the find tool in the editor and replace all instances of the following with your information. ##VOIPMS## - Replace with your voip.ms server of choice. ##LABEL## - this is the label that appears on the phone screen beside the line. You can put whatever you want here. ##VOIPMSUSER## - this is your voip.ms username number. ##VOIPNAME## name you want to be sent to VOIP.ms. Not sure what this is for but you can safely put your name such as John here. ##VOIPMSPASS## - your voip.ms password. Be sure to set this in your account settings. Save the file in your cisco folder.
Step 4) Download the dialplan.xml I provided. No need to edit it. Place it in the cisco folder.
Step 5) reboot your phone (settings button then * * # * *) With luck, your phone should reboot and download the xml files in the cisco folder and register with voip.ms. Congrats!
BONUS INFO!
Follow this URL to learn how to install custom backgrounds and other cool things with the phone. https://www.whizzy.org/2017/02/cisco-7941-asterisk-and-sip/#flash
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Obihai OBi200/202/302 + OBi1022/1032/1062 + OBi50x firmware mods
So I want to add the ability to configure these devices for GV using oauth without obitalk, similar to the changes for the obi100 (and add an ssh server, for grins).
I think I have the MD5s in the firmware file worked out (its the same "Goodbye! Reboot Now" garbage as the 100), and I see where the oauth refresh token code is, so it should be pretty straightforward unless there is code signing that I missed.
The only hiccup is... I don't actually have an OBi20x :-(
Anyone have one of these devices that wants to be a guinea pig? You should definitely have a way to SPI the flash back *when* i brick the thing the first couple tries...
[or if someone has one sitting in the closet, you could just send it to me. ill name the fw after you :-)]
EDIT: speaking of flash, its supposed to have a w25x128 on board, but is it the SOIC package or some BGA madness?
QUICK SUMMARY:
Custom firmware made for all obi devices, thanks to the help of generous hardware donations and bold testers.
See obifirmware.com to download latest.
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VOIPo down again
I know, I sound like a broken vinyl record. Both 618-632 #### and 618-624 #### lines are down and I guess they were down yesterday too but I did not follow-up on wife's complaint yesterday. Anyway today incoming rings after calling party's call already went to voicemail. Outbound calls do not connect but sorta ring the calling part with one ring but no connections. Courtesy ringback generally does not work.
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X-Lite 3: syntax of remote_party_id_preference_order?
In X-Lite 3 Advanced Options (***7469) there's an option named remote_party_id_preference_order, whose value is blank. I assume this is a list of identity headers in the SIP INVITE to check for caller identity. Does anyone know the syntax for the value of this option? I already searched the Counterpath knowledge base and tried Google Search. I haven't experimented yet.
As far as I can tell there is no documentation for these Advanced Options.
Thanks!
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VOIPo Still down
Inbound kept going to cell phone failover this morning, I had been warned to use the VOIPo ATA so I reluctantly switched to the Voipo ATA HT502 and it looks fine in VPanel connected devices but still does not work. The symptoms are too bizarre to describe. The other VOIPo line at the at the American Legion appears to be down too as I get the rejection recording reason 408 when called from a non-VOIPo phone. Both lines are on the VOIPo ATA. So much for that remedy.
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[Equipment] Google Fi data-only nano SIM softphone...
Somebody may find this interesting... I'm still in the trial stage... so far it's working...
My wife has Google Fi mobile service (uses Sprint, T-Mobile, and US Cellular).
Google Fi will ship her free data-only nano SIMs to enable additional devices with T-Mobile data-only service at the going Google Fi data rate (I have seen it connect to US Cellular, but did not have a chance to use it). These devices do not require her Google account on them.
So, I put a GFi data-only nano SIM in a spare/reset Samsung Galaxy S6 phone (was US Cellular), installed Zoiper, and will see how it runs and what it costs for my usage.
(One pleasant surprise was discovering that the Samsung Galaxy S6 supports Qi wireless charging.)
OE
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[Anveo] Modify voicemail profile?
I have a couple of different voicemail profiles set up in my call flow. I want to modify one of them, but I can't figure out how to do that.
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[Voip.ms] New Beta feature - IP & POP Restriction
Just received an email from voip.ms about this new feature:
"...we would like to notify you of the start of the beta-testing phase of two new features: IP restriction and POP restriction.
IP Restriction: This feature is located under “Main Menu”, “Account Settings” then under “Security” tab. When enabled, outgoing calls will be authorized only from the IP address or IP range specified.
POP Restriction: This feature is also located under “Main Menu”, “Account Settings” then under “Security” tab. When enabled, outgoing calls will be authorized only from the Points of Presence (POPs) selected.
Please note that for both features, you may also easily replicate the restriction to all your sub accounts at once or do it individually by heading over to the sub accounts menu. For more information, you may consult our Wiki article here."
I guess this is is meant to minimize the fraud attempts similar to the ones that users were complaining recently in this forum.
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Busy signal
While I am new to voip.ms, I am not new to VOIP and have set up several ATAs. I created a voip.ms account along with sub account. Both register just fine and I can call out from both lines. The problem I have having is that when I call either DID number I get a busy signal and the phone does not ring. Any suggestions?
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