This mysterious hacking campaign snooped on a popular form of VoiP software A hacking campaign is targeting one of the world's most popular services for making voice over IP phone calls, allowing the attacker to snoop on who people are calling and when they're calling them, listen to recordings of conversations and send out spoof calls that look like they come from the legitimate number of the compromised user. The attack has been detailed during a presentation by Check Point researchers at the Virus Bulletin 2019 conference in London. Security researchers have traced the initial attacks back to between February and July 2018, when an attacker was performing scans on over 600 companies across the world that use Asterisk FreePBX – a popular form of open source VoiP software. The attacker then went quiet for months before re-emerging this year, targeting a US-based server owned by an engineering company that provides services to the oil, gas and chemical industries.... MORE: https://www.zdnet.com/article/this-mysterious-hacking-campaign-is-snooping-on-a-popular-form-of-voip-software/ Did we miss this? Story was from few weeks ago (October 4, 2019).

My business has had an 888 number for 10, maybe 15 years. The TN hasn't changed. It was ported over to voip.ms about 3 years ago, same time as a couple of the business's other POTS numbers were ported. The 888 number is not listed anywhere, we use it when a client has to call us but his institution can't call long distance (or can't call Canada) and we also used to use it when we were out of the office and need to call back to the office (from a pay phone, etc). To be honest, it probably hasn't been used (for a legit call) for the past 2 years. I noticed today that a single call came in to the 888 number in each of July, August and September. Two of them were from the 606 area code, the other was 678. Each call was under a minute, either not long enough to ring through to our reception line or if it did, no message was left or nobody was there when the phone was answered. Two of these calls have CID saying simply "Wireless Caller", the other says "Jeptha, KY". Now, starting Oct 4, I see 7 calls to my 888 number. Oct 4, 14, then 23 to 27 (one each day). 206, 608, 816, 518, 907, 706, and today 419 area code. Calls range from 46 seconds to almost 2 minutes. A human mis-dialling our number would hear our phone system greeting and realize they've called the wrong number within 10 seconds. I've google'd our 888 number to see if someone else has mistakenly put it up somewhere as their number, but nothing turns up. I happened to be at my office when this last one came in today, and it rang through to reception and I picked up, said hello several times, held the line open for maybe 20 seconds, but just got silence. About a minute later I called the number back, and it's a business in Ohio (I get same info when I google the number). Nobody answered - I got their voice mail. I think I'll call them tomorrow and ask if they called my 888 number - on a Sunday at 6 pm. These calls are coming in anywhere from 11:30 am to 9 pm (EST) except one happened at 2 am. Is it worth it to take this up with voip.ms? Given that this is a toll-free DID, will they be able to tell me anything about who is really calling beyond what I'm seeing in the call records? Or can I assume that the numbers I'm seeing are spoofed, even if the toll-free "system" is supposed to drill through and give me (the guy being billed for these calls) the "real deal" when it comes to caller ID. ? Anyone else seeing this on their toll-free did's ?

Would one work with my Obi ATA? I want the kids to enjoy the feel of the old way of dialing.

Seems as if it was only a few weeks ago when it was out. Now it's down again. No inbound, no outbound calls getting through. Their twitter page confirms they are having issues.

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

Hello Guys I have an spa122 haves this as dial plan "(*xx|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxx.)" I dont know if that is correct dial plan for my free pbx I am using or not.

I have been using bulkvs.com for VoIP 911 services for a couple of years. I've now expanded to use all their services. Wow! It's great. IP or registration authentication. Origination and Termination. Tech support is VERY fast in responding to issues -- usually resolved within an hour or less. I've ported a ton of DIDs to them -- painless process, very fast. $0.0055 TF Inbound rate -- lowest I can find, most are 1.5- - 1.9/c/min (Canada/USA) $0.0003 Inbound/$0.004 outbound prices for regular SIP traffic is among the lowest I've found with the low MRN. No MINIMUMS! They pay you if you route outbound TF traffic through them. I am just a customer and a fan. YMMV. Check em out. Sample of rates (this is my quote, but not significantly diffeernt from public): Inbound Location Service Monthly Rate/Min LNP Fee DID US-48 Tier 0 $0.06 $0.0003 $0.00 DID Tier 1 $0.15 $0.0065 $0.00 DID Tier 2 $0.15 $0.0099 $0.00 DID Tier 3 $0.15 $0.0171 $0.00 DID Tier 5 $0.25 $0.0198 $0.00 Toll Free US-48 / Canada $0.14 $0.0055 $0.00 Toll Free Alaska $0.14 $0.030 $0.00 Toll Free Hawaii $0.14 $0.009 $0.00 Toll Free Guam $0.14 $0.019 $0.00 Toll Free U.S. Virgin Islands $0.14 $0.0095 $0.00 Additional Services Product Rate/Cost CNAM $0.002 LRN $0.0002 Messaging SMS Enabling $0.01 Messaging SMS Inbound / Outbound $0.004 / $0.004 E911 per number $0.49 E911 Overage Fee $5 E911 Unprovisioned calls $85 LIDB Updates $0.10 Directory Listing $0.99 Toll Free Vanity Number $0.00 Outbound Service Rate/Min Outbound Calling $0.004 Toll Free (We credit your account) $0.0012 - 1 to 99,999 minutes $0.0016 - 100,000 to 999,999 minutes $0.0020 - 1,000,000+ minutes

have an anveo account and would like to use my laptop and headphones to accept and make calls. whats a good software to use for this? i have multiple dids with anveo and is there a way to have when certain calls come in have previous input data pop up on screen. for example if someone dials a particular DID of mine and it rings on my end, i have a batch number that i need to know for that DID. right now i just go through a rolodex to get that number. i would like to add some code in the call flow that will auto pop up that batch number on the screen when the calls comes in. I'll figure out the code later just want to make sure what software i use will support that. TIA

California’s recent power shut-offs, meant to reduce the risk of potentially catastrophic fires, have had an unwelcome side effect. The blackouts have also cut power to many cellphone towers, blocking the main communications source for many in harm’s way. “You don’t appreciate how essential cellphone service is until you lose it,” said Chris Ungson, deputy director for communications and water policy for the California Public Advocates Office, an independent agency within the state’s Public Utilities Commission. “It’s not just a matter of inconvenience; it’s a matter of public health and safety. It’s a lifeline to many, many people.” Emergency calls to 911 are one indicator: The Governor’s Office of Emergency Services said more than 80 percent of such calls in California last year were made by cellphone. For years, state and federal regulators have pressed the cellular companies to better reinforce their networks for emergencies. The Federal Communications Commission said Monday that it was conducting “a comprehensive review of the wireless industry’s voluntary commitment to promote resilient wireless communications during disasters.” https://www.nytimes.com/2019/10/28/business/energy-environment/california-cellular-blackout.html Cellphone service has a lot in common with VoIP services, both are dependent on reliable ongoing electric supply. A telecom ecosystem that included some legacy landline service as one of its components, would have been advantageous for the country. We're probably past the point of no return on that, though.

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.

Have several DIDs in the 914 area code with CC. Noticed this afternoon calls were not coming through even when directly dialed from a cell phone/other voip provider (no gv involved). Anyone else experiencing similar?

Greetings, I think I'm interested in VOIP for my residence and I'm not too knowledgeable about it other than it's a phone service operated via the Internet and it is potentially cheaper. I currently have cable Internet and phone service via Suddenlink. And the Caller ID service does not work for me and seems that SL can't fix it and I need for the Caller ID service to work for phone line monitoring for scam filtering. I think I want to change so I can save money and have more calling features and for better scam filtering. In the meantime I will educate myself more about VOIP. What are the better/best services available for residential use and what is involved in changing over to VOIP? What is the general amount of time needed to port a phone number? Do most voip services include e911 services? I'm interested in what you have tried and did it work best. I'm looking for some general information that I can use that will help to guide me in the right choice to make. Any good websites for a newbie to use such as myself? Thanks for your time and suggestions.

Anyone else having problems making and receiving calls? Both my home phone and the one downtown at our American Legion post are not working and this occurred yesterday too. Earlier today, the home phone was accepting in coming calls. The VOIPo reliability has really reached unacceptable levels. NPA/NXX are 618-632 and 618-624.

very easy to install, but difficult to understand dsiprouter's GUI , difficult for new user. I believe need much time to learn it and intent to quit it for time-efficiency. tried various IP PBX in past ten years, still love trixbox and Elastix, but both are gone (don't talk about the 3CX). Now running issable on PC, very stable and easy to use. recommended. Unfortunately Issabel doesn't support ARM cpu, so run Freepbx and FusionPBX on raspberry and orange pi, but like FusionPBX more because very easy to install FusionPBX, just two line commands, and most modules are free. Rank them as below: 1. Issabel 2. FusionPBX 3. FreePBX 4. ? 5. ? 6. ? .......................................

Alright so I created a thread a 2 year ago asking for telephone equipment that supports sidetone. At the time, I was looking for a PSTN/ Telephone edition. Now I'm looking for the mobile / cellular version. This means 3.5mm, or USB OTG, or Bluetooth. The person is losing their hearing and yells on the phone because they can't hear themselves speak. It is driving other people in the home insane. Money is no object here. The hearing loss is not yet severe and should not require medical grade levels of boost. however, if that happens to be the best available equipment, I'm onboard. I definitely don't want to skimp on boosting range as I don't know how much boost will be required for the person to change their behavior. Heck, human factors suggestions are welcome too. I'm guessing the optimal solution is a normal speaker volume with high quality (de-emphasises remote party's volume level to discourage competing) and a boosted sidetone volume (emphasizes that the other person can hear you loud and clear) Requirements: - Must offer a high volume sidetone option that does not degrade the calling experience. - Must be compact and comfortable enough the user will be encouraged to use it. - Bukly full-sized headphones are also potentially fine, as long as they allow freedom of movement and comfort. - Must offer decent battery life, if required. - If Bluetooth, must offer Wideband audio calling (such as the trekz aftershockz air), which offer excellent call quality. - Should probably feature a boom microphone for optimal call quality

I recently did a fresh install of Asterisk 16.5 and Debian 9.8 on my Beaglebone black. Been running 16.1 on Debian 8.4 for quite a while with no issues. Shortly after the upgrade, I was talking on a toll free number using Flowroute and about 8 minutes into the call it just dropped. My sister has used Flowroute with an Obi 100 for years and I have used them with Asterisk off and on for quite a while. I realize one dropped call does not constitute a fire drill but have been reading lots of "instances" of questions about late-versions of Asterisk and various PBXes so am wondering if I should drop back to 16.1 or maybe it could be the OS?

Here is an 'easy' install of naf Asterisk (aka GVsip). There is no GUI, I prefer it this way. I have pre-configured it for up to 10 GV accounts (except for personal info). You can easily add more. You will need oAuth2 credentials for each GV account. For hardware I used: An old Core 2 Duo (2008) with 2GB ram and a 20GB hard drive. It uses 50 Watts continuous power. If you have a newer computer it will probably use less power. A newer Celeron would be great (under 15 Watts). Connect the ethernet port of the soon-to-be Ubuntu/Asterisk server to your router. Connect the Obi100 to the same router. I used a DVD with Ubuntu Server 17.1 to do a fresh install and then: sudo do-release-upgrade (installs latest Ubuntu version 18.04.1) sudo apt install gcc (installs C compiler) On your regular computer: Download the ZIP file I posted and extract the four configuration files. Load pjsip.conf into Notepad (or similar). Choose which SIP account (1001, 1002... through 1010) you'll be using. You'll only be editing in that account. Where it shows 'agoodpassword' change this to a made-up password you'll use later in the Obi. Enter your Google Voice 10 digit phone number in place of 1112223333 in TWO places. Add your oAuth credentials. Use copy and paste to avoid typing errors. Don't leave any spaces before or after. In place of 444555666 enter your Obi number (written on the bottom of the Obi). You can make up a number if you wish of around 9 digits. Save the file (without any formatting). Copy the newly edited file and the other three .conf files to a USB stick. Then back to the Ubuntu computer: cd /usr/srcsudo git clone https://github.com/asterisk/asterisk.gitcd asterisksudo contrib/scripts/install_prereq install (runs the installation script) (It may pop up with "ITU-T Telephone Code", enter 1 for United States.)sudo ./configure ;sudo make menuselect (optional)sudo make (compiles C into objects)sudo make install (links objects, downloads stuff)sudo make config (Configure as service at bootup)sudo cp configs/samples/*.* /etc/asteriskcd /etc/asterisksudo find . -name "*.sample" -exec sh -c 'mv "$1" "${1%.sample}"' _ {} \; ;Insert your USB stick with the four .conf files.lsblk (this will list drives. The USB will probably be sdb1)sudo mkdir /usb (make a USB directory (on HDD) to mount the USB drive)sudo mount /dev/sdb1 /usb (USB contents are now available in /usb)sudo cp /usb/*.conf /etc/asterisk (copy our .conf files)sudo umount /dev/sdb1 (log out of USB stick) ;remove your USB stick reboot At this point Asterisk will be running in the background and your Obi will connect if you have already configured it. Tips on using a running Asterisk: sudo asterisk -r (continues running but also gives you the Asterisk CLI (command line). core stop now (stops Asterisks if you need to make configuration changes.) sudo nano /etc/asterisk/pjsip.conf (edit the pjsip.conf file) sudo asterisk -cvvvv (re-starts Asterisk with verbose level 4) Configuring an Obi100 (other ATA's, IP phones, and softphones may be similar): Open a browser on a computer on the same LAN and log into your router to find the IP address of the Ubuntu/Asterisk server and the Obi100. Log into the Obi and go to Service Providers, ITSP Profile A, SIP Uncheck ProxyServer and type in the address of the Asterisk server. Something like 192.168.1.5 Uncheck ProxyServerPort and type in 5085 Scroll down to the bottom and click on Submit (and OK). Go to RTP (on the menu). Enable RTCP and X_RTCPMux Click on Submit and OK. Go to Voice Services, SP1 Service, SIP Credentials, AuthUserName, 1001 or whatever account # you configured earlier. AuthPassword, Enter what you used in place of 'agoodpassword'. Scroll down to bottom and click on Submit and OK. Click on Reboot in the upper right. It should register with the Asterisk server indicated by the phone LED (on the Obi) lighting up. Quick Links: Install in Mint-Cinnamon https://www.dslreports.com/forum/r32078999-How-to-Install-naf-Asterisk-on-Ubuntu-for-Obi100-and-Google-Voice~start=60 Install in old laptop and in a VM https://www.dslreports.com/forum/r32078999-How-to-Install-naf-Asterisk-on-Ubuntu-for-Obi100-and-Google-Voice~start=56 Install in Debian-Cinnamon https://www.dslreports.com/forum/r32078999-How-to-Install-naf-Asterisk-on-Ubuntu-for-Obi100-and-Google-Voice~start=127 Install in Windows 10 https://www.dslreports.com/forum/r32078999-How-to-Install-naf-Asterisk-on-Ubuntu-for-Obi100-and-Google-Voice~start=391 naf donated certificates https://www.dslreports.com/forum/r32016984-Asterisk-Google-Voice-SIP-testing-and-technical-discussion~start=1151 Certificate Extraction https://www.dslreports.com/forum/r31741105-ObiHAI-Obi100-Obi110-Firmware-Mod-Discussion~start=653 Obi200+ Now Required https://www.dslreports.com/forum/r32078999-How-to-Install-naf-Asterisk-on-Ubuntu-for-Obi100-and-Google-Voice~start=330 Adding Google Contacts https://www.dslreports.com/forum/r32078999-How-to-Install-naf-Asterisk-on-Ubuntu-for-Obi100-and-Google-Voice~start=38 Info on *67 (block your caller id) https://www.dslreports.com/forum/r32078999-How-to-Install-naf-Asterisk-on-Ubuntu-for-Obi100-and-Google-Voice~start=42 Spam blocker https://www.dslreports.com/forum/r32078999-How-to-Install-naf-Asterisk-on-Ubuntu-for-Obi100-and-Google-Voice~start=180 Asterisk/GVsip information https://github.com/naf419/asterisk/wiki Edit1: Github file location for Asterisk New configs.zip file. Has new versions of pjsip.conf and extensions.conf. Edit2: New configs.zip file. Has new versions of pjsip.conf and extensions.conf. Edit3: Added 'Quick Links' above to jump to pertinent info. New configs.zip file. Has new versions of pjsip.conf, extensions.conf, and modules.conf. Edit4: New "method" param added to pjsip.conf under transport_tls to match pjproject change made in Asterisk past 6-27-2019. Updated extensions.conf file.

I managed to find a Cisco 7942 phone on sale for $15. I had always wanted a Cisco phone to use at home. So I bought it! Little did I know that these phones are not configured to be used with SIP out of the box. They use a proprietary protocol called Skinny Client Control Protocol (SCCP). The phone was basically useless. Or was it? In this post, I'll be posting instruction on how I successfully converted my Cisco 7942 phone from the SCCP protocol to SIP **AND** got it to work with VOIP.ms. The system I am used to get the phone working is Linix (Debian), however you should have no problem getting this to work on a Mac or Windows computer simple by using different programs than I ones I talk about here. Here is a summary of the steps I used to complete the process. I will elaborate on the details below: 10 Steps to Convert phone from SCCP protocol to SIP protocol Step 1 - Install a TFTP server Step 2 - Configure TFTP server Step 3 - Install a DHCP server Step 4 - Configure DHCP server Step 5 - Download SIP firmware cmterm-7942_7962-sip.9-4-2SR3-1.zip Step 6 - Configure XMLDefault.cnf.xml file Step 7 - Configure dialplan.xml Step 8 - Configure SEPxxxx.cnf.xml file Step 9 - Preform a hard reset on the Cisco 7942 phone Step 10 - Start making calls ------------------------------ Step 1 - Install a TFTP server ------------------------------ This step will show you how to install a TFTP server on Linux (Debian). If you are using Windows or a Mac, install the TFTP software specific to your system. With linux, you need to installed three programs. xinetd to control the services. tftpd to install the tftp server. and tftp for a generic tftp prompt. You can use the following prompt: $ sudo apt-get install xinetd tftpd tftp ------------------------------- Step 2 - Configure TFTP server ------------------------------- Now that you have the tftp server installed, you will need to create a directory that you will be placing all of your files into and permissioning the directories accordingly. If you are using Windows or Mac, you can edit these settings via the GUIs settings. On Linux, you can use the following commands: $ sudo mkdir /tftpboot $ sudo chmod -R 777 /tftpboot $ sudo chown -R nobody /tftpboot Now that you have a directory to hold all of your files, you need to configure your tftpd config file. Edit the config by using he following command: $ sudo nano /etc/xinetd.d/tftp and replace all of the existing code with the following: X service tftp { protocol = udp port = 69 socket_type = dgram wait = yes user = nobody server = /usr/sbin/in.tftpd server_args = /tftpboot disable = no } Now we will test the tftp to see if it works. If this part below fails, go back to the beginning of Step 2 and try again. Start the tftp server: $ sudo service xinetd restart $ sudo service xinetd stop $ sudo service xinetd start First we will make a dummy file called 'test' and place it in the tftp server directory $ ls / > /tftpboot/test Next we try to connect to the tftp server. In my case, the ip number of the computer is 192.168.1.10. Change this to whatever IP number your router passed onto your computer. $ tftp 192.168.1.10 Now we will try to download the 'test' file to see if our tftp server works: tftp> get test You should see something similar to: Sent 159 bytes in 0.0 seconds as the result. If you do, simply quit the program by typing in: tftp> quit and move to Step 3. If any part above fails, start over with Step 2. ------------------------------ Step 3 - Install a DHCP server ------------------------------ Just like above, this step shows how to install a DHCP server on Linux (Debian). If you are using Windows or a Mac, install the DHCP software specific to your system. You can use the following prompt: $ sudo apt-get install isc-dhcp-server ------------------------------- Step 4 - Configure DHCP server ------------------------------- Edit the DHCP config file by using the following command: $ sudo nano /etc/dhcp/dhcpd.conf Replace all of the contents of the config file with the following: X # Do not add an IP address below. Leave it as '{ ip-address }' option voip-tftp-server code 150 = { ip-address }; # Make the ip number below whatever ip the DHCP computer will be on option voip-tftp-server 192.168.2.10; # Use Google public DNS server (or use faster values that your internet provider gave you!): option domain-name-servers 8.8.8.8, 8.8.4.4; # Set up our desired subnet: subnet 192.168.2.0 netmask 255.255.255.0 { range 192.168.2.101 192.168.2.254; option subnet-mask 255.255.255.0; option broadcast-address 192.168.2.255; option routers 192.168.2.1; option voip-tftp-server 192.168.2.10; } default-lease-time 600; max-lease-time 7200; # Show that we want to be the only DHCP server in this network: authoritative; And start the DHCP server by using the following command: $ sudo service isc-dhcp-server start Don't forget to shut off the DHCP server on your router before proceeding to the next step ------------------------------------------------------------------ Step 5 - Download SIP firmware cmterm-7942_7962-sip.9-4-2SR3-1.zip ------------------------------------------------------------------ This will be the trickiest part. As the firmware is considered proprietary, I can not provide a direct link to the site. The latest firmware zipfile is called cmterm-7942_7962-sip.9-4-2SR3-1.zip. You can do a Google search for that file or search for cisco 7942 sip firmware download. Unzip all of the files contained within this file into your TFTP server directory. In my case I would copy them to /tftpboot Now this is probably the most important piece of this entire document. Whatever version of the firmware you get, there is a file called SIPxxxx.loads in the directory. The xxxx is whatever firmware you are using. In my case, the .loads file I have is called SIP42.9-4-2SR3-1S.loads. Whatever the file name is called **BEFORE** the .loads write it down and remember it. You will need to use this **exact** sequence later in the .xml files. In my case, I will write down and remember SIP42.9-4-2SR3-1S ------------------------------------------- Step 6 - Configure XMLDefault.cnf.xml file -------------------------------------------- Create a file called XMLDefault.cnf.xml and place it in the TFTP server directory. Copy and paste the lines below into that file. Keep in mind that the <loadInformation434>SIP42.9-4-2SR3-1S</loadInformation434> is a very important line. Whatever the name of the firmware above is you will need to ensure that same wording is between the loadinformation tags. X <Default> <callManagerGroup> <members> <member priority="0"> <callManager> <ports> <ethernetPhonePort>2000</ethernetPhonePort> <mgcpPorts> <listen>2427</listen> <keepAlive>2428</keepAlive> </mgcpPorts> </ports> <processNodeName></processNodeName> </callManager> </member> </members> </callManagerGroup> <loadInformation434>SIP42.9-4-2SR3-1S</loadInformation434> <authenticationURL></authenticationURL> <directoryURL></directoryURL> <idleURL></idleURL> <informationURL></informationURL> <messagesURL></messagesURL> <servicesURL></servicesURL> </Default> -------------------------------- Step 7 - Configure dialplan.xml -------------------------------- Create a file called dialplan.xml in your tftp directory. You can use your own dialplan or you can copy and paste what I have below. This is up to you: X <DIALTEMPLATE> <TEMPLATE MATCH="10." Timeout="0" User="Phone"/> <!-- Access to extensions --> <TEMPLATE MATCH="\*97" Timeout="0"/> <!-- Voicemail --> <TEMPLATE MATCH="1.........." Timeout="0" User="Phone"/> <!-- Immediate dialout out to long distance --> <TEMPLATE MATCH=".........." Timeout="0" User="Phone"/> <!-- Immediate dialout out to local --> <TEMPLATE MATCH="*" Timeout="5" User="Phone"/> </DIALTEMPLATE> ---------------------------------------- Step 8 - Configure SEPxxxx.cnf.xml file ---------------------------------------- Before proceeding with Step 8, you need to know the MAC address of your Cisco 7942 IP Phone. To find this out, flip the phone over and write down the MAC address on a piece of paper. Lets pretend the MAC address of my phone is 123ABC789XYZ. I will create a file in tftp server directory called SEP123ABC789XYZ.cnf.xml and add the following lines to it. You will see some comments within the XML file. Be sure to abide by them. Also just like the XMLDefault file above, you need to have the firmware name in the <loadInformation>section. X<device><deviceProtocol>SIP</deviceProtocol> <sshUserId>cisco</sshUserId> <sshPassword>1234</sshPassword> <devicePool><dateTimeSetting> <dateTemplate>D/M/Y</dateTemplate> <timeZone>Eastern Standard/Daylight Time</timeZone> <ntps> <ntp> <name>208.73.56.29</name> <!-- Only an IP # for a NTP server may be used --> <ntpMode>Unicast</ntpMode> </ntp> </ntps></dateTimeSetting> <callManagerGroup> <members> <member priority="0"> <callManager> <name>VOIPMS</name> <!-- Name of Voip Provicer --> <ports> <ethernetPhonePort>2000</ethernetPhonePort> <sipPort>5060</sipPort> <!-- Voip Port To Use --> <securedSipPort>5061</securedSipPort> <mgcpPorts> <listen>2427</listen> <keepAlive>2428</keepAlive> </mgcpPorts> </ports> <processNodeName>toronto1.voip.ms</processNodeName> <!-- Voip.ms Server Address --> </callManager> </member> </members></callManagerGroup></devicePool><sipProfile><sipProxies><backupProxy /> <backupProxyPort /> <emergencyProxy /> <emergencyProxyPort /> <outboundProxy /> <outboundProxyPort /> <registerWithProxy>true</registerWithProxy> <!-- Must be set to TRUE for use with Voip.MS --></sipProxies><sipCallFeatures> <cnfJoinEnabled>true</cnfJoinEnabled> <callForwardURI>x-serviceuri-cfwdall</callForwardURI> <callPickupURI>x-cisco-serviceuri-pickup</callPickupURI> <callPickupListURI>x-cisco-serviceuri-opickup</callPickupListURI> <callPickupGroupURI>x-cisco-serviceuri-gpickup</callPickupGroupURI> <meetMeServiceURI>x-cisco-serviceuri-meetme</meetMeServiceURI> <abbreviatedDialURI>x-cisco-serviceuri-abbrdial</abbreviatedDialURI> <rfc2543Hold>false</rfc2543Hold> <callHoldRingback>2</callHoldRingback> <localCfwdEnable>true</localCfwdEnable> <semiAttendedTransfer>true</semiAttendedTransfer> <anonymousCallBlock>0</anonymousCallBlock> <callerIdBlocking>0</callerIdBlocking> <dndControl>1</dndControl> <remoteCcEnable>true</remoteCcEnable> </sipCallFeatures> <sipStack> <sipInviteRetx>6</sipInviteRetx> <sipRetx>10</sipRetx> <timerInviteExpires>180</timerInviteExpires> <timerRegisterExpires>180</timerRegisterExpires> <timerRegisterDelta>5</timerRegisterDelta> <timerKeepAliveExpires>120</timerKeepAliveExpires> <timerSubscribeExpires>120</timerSubscribeExpires> <timerSubscribeDelta>5</timerSubscribeDelta> <timerT1>500</timerT1> <timerT2>4000</timerT2> <maxRedirects>70</maxRedirects> <remotePartyID>false</remotePartyID> <userInfo>None</userInfo> </sipStack><autoAnswerTimer>1</autoAnswerTimer> <autoAnswerAltBehavior>false</autoAnswerAltBehavior> <autoAnswerOverride>true</autoAnswerOverride> <transferOnhookEnabled>false</transferOnhookEnabled> <enableVad>false</enableVad> <preferredCodec>g711ulaw</preferredCodec> <dtmfAvtPayload>101</dtmfAvtPayload> <dtmfDbLevel>3</dtmfDbLevel> <dtmfOutofBand>avt</dtmfOutofBand> <alwaysUsePrimeLine>true</alwaysUsePrimeLine> <alwaysUsePrimeLineVoiceMail>false</alwaysUsePrimeLineVoiceMail> <kpml>3</kpml> <natEnabled>true</natEnabled> <natAddress>123.456.789.123</natAddress> <!-- Replace this IP address with your external IP address --> <phoneLabel>VOIP Phone</phoneLabel> <!-- Maximum 12 Characters Message To Appear On Your Phone--><stutterMsgWaiting>1</stutterMsgWaiting> <callStats>false</callStats> <silentPeriodBetweenCallWaitingBursts>10</silentPeriodBetweenCallWaitingBursts> <disableLocalSpeedDialConfig>false</disableLocalSpeedDialConfig> <startMediaPort>16384</startMediaPort> <stopMediaPort>32766</stopMediaPort> <sipLines><line button="1"> <featureID>9</featureID> <featureLabel>Line 101</featureLabel><!-- Display name of button --> <proxy>USECALLMANAGER</proxy><!-- Must use call manager - do not enter VOIP servers here --> <port>5060</port> <name>111222</name><!-- This is the Voip.ms USERNAME It must be the same as<authName> below --> <displayName>Billy Bob</displayName><!-- Caller ID CNAM --> <autoAnswer> <autoAnswerEnabled>0</autoAnswerEnabled> </autoAnswer> <busyTrigger>1</busyTrigger><!-- Number of calls the line can have at a time before returning a busy signal. --> <authName>111222</authName><!-- This is the Voip.ms USERNAME It must be the same as<name> above --> <authPassword>abc123</authPassword> <!-- This is the Voip.ms USERNAME password --> <sharedLine>false</sharedLine> <messageWaitingLampPolicy>3</messageWaitingLampPolicy> <messagesNumber>*97</messagesNumber><!-- Phone number to dial when VM button is pressed --> <ringSettingIdle>4</ringSettingIdle> <ringSettingActive>5</ringSettingActive> <contact></contact> <forwardCallInfoDisplay> <callerName>true</callerName> <callerNumber>true</callerNumber> <redirectedNumber>false</redirectedNumber> <dialedNumber>true</dialedNumber> </forwardCallInfoDisplay></line><line button="2"> <featureID>9</featureID> <featureLabel>Line 102</featureLabel><!-- Display name of button --> <proxy>USECALLMANAGER</proxy><!-- Must use call manager - do not enter VOIP servers here --> <port>5060</port> <name>11234</name><!-- This is the Voip.ms USERNAME It must be the same as<authName> below --> <displayName>Billy Bob</displayName><!-- Caller ID CNAM --> <autoAnswer> <autoAnswerEnabled>0</autoAnswerEnabled> </autoAnswer> <busyTrigger>1</busyTrigger><!-- Number of calls the line can have at a time before returning a busy signal. --> <authName>11234</authName><!-- This is the Voip.ms USERNAME It must be the same as<name> above --> <authPassword>556677!</authPassword> <!-- This is the Voip.ms USERNAME password --> <sharedLine>false</sharedLine> <messageWaitingLampPolicy>3</messageWaitingLampPolicy> <messagesNumber>*97</messagesNumber><!-- Phone number to dial when VM button is pressed --> <ringSettingIdle>4</ringSettingIdle> <ringSettingActive>5</ringSettingActive> <contact></contact> <forwardCallInfoDisplay> <callerName>true</callerName> <callerNumber>true</callerNumber> <redirectedNumber>false</redirectedNumber> <dialedNumber>true</dialedNumber> </forwardCallInfoDisplay></line></sipLines><voipControlPort>5060</voipControlPort> <!-- Port # to Use --><dscpForAudio>184</dscpForAudio> <ringSettingBusyStationPolicy>0</ringSettingBusyStationPolicy> <dialTemplate>dialplan.xml</dialTemplate> </sipProfile><commonProfile><phonePassword /> <backgroundImageAccess>true</backgroundImageAccess> <callLogBlfEnabled>3</callLogBlfEnabled> </commonProfile><loadInformation>SIP42.9-4-2SR3-1S</loadInformation> <!-- Firmware name to go here --><vendorConfig><disableSpeaker>false</disableSpeaker> <disableSpeakerAndHeadset>false</disableSpeakerAndHeadset> <pcPort>0</pcPort> <settingsAccess>1</settingsAccess> <garp>0</garp> <voiceVlanAccess>0</voiceVlanAccess> <videoCapability>0</videoCapability> <autoSelectLineEnable>0</autoSelectLineEnable> <webAccess>0</webAccess> <spanToPCPort>1</spanToPCPort> <loggingDisplay>1</loggingDisplay> <loadServer /> </vendorConfig><versionStamp>1143565489a3cbf29475264c298791c4fce4ce4c37</versionStamp> <networkLocale>US</networkLocale> <networkLocaleInfo><name>US</name> <version>5.0(2)</version> </networkLocaleInfo><deviceSecurityMode>1</deviceSecurityMode> <authenticationURL /> <directoryURL></directoryURL> <idleURL /> <informationURL /> <messagesURL /> <proxyServerURL>proxy:3128</proxyServerURL> <servicesURL /> <dscpForSCCPPhoneConfig>96</dscpForSCCPPhoneConfig> <dscpForSCCPPhoneServices>0</dscpForSCCPPhoneServices> <dscpForCm2Dvce>96</dscpForCm2Dvce> <transportLayerProtocol>2</transportLayerProtocol> <capfAuthMode>0</capfAuthMode> <capfList><capf><phonePort>3804</phonePort> </capf></capfList><certHash /> <encrConfig>false</encrConfig> </device> ------------------------------------------------------ Step 9 - Preform a hard reset on the Cisco 7942 phone ------------------------------------------------------ Power down the phone. Turn the phone back on and push and continue holding the # key while its booting up until the two lights on the display panel start to flash. Once the lights are flashing, release the # key and push the following buttons: 123456789*0# From here the phone will take over the controls and connect to the tftp server. It will flash the firmware on the phone to allow you to use the SIP protocol. It will also install your dialplan and configuration files for the phone. This process should take about 5 - 10 minutes to complete. ---------------------------- Step 10 - Start making calls ---------------------------- The phone should be registered and you'll be all set. If for whatever reason your phone is not registerting with VOIP.ms, here are some troubleshooting steps: 1) Enable portforwarding within your router to allow the IP Phone to have access to ports 5060-5080 2) Under Voip.ms Account Settings > Advanced. If NAT (Network Address Translation) is set to NO, change it to YES. Or vice versa 3) Go back to Step # 6 and ensure the settings for the SEPxxxx.cnf.xml are correct. Make sure <natAddress> reads as your CURRENT EXTERNAL IP ADDRESS. If your IP # changes, you will have remake the config file. Make sure that the Username's server address on VOIP.ms matches the server name in the <processNodeName> tag. The <phoneLabel> tag can only have a maximum of 12 character. If you add more the phone will not register. The <name> and <authname> tags should be identical and read as your VOIP.ms username. Once you are done everything here, you will turn on the DHCP server on your router and shut off this DHCP service with the following command: $ sudo service isc-dhcp-server stop You're all done. Enjoy your calls.

I'm dusting off a working Grandstream HT701 ata with firmware 1.0.6.1. I flashed it to the latest firmware 1.0.9.4 but then cannot get it to register with VoIP.ms again using the same settings. The firmware change notes since 1.0.6.1 are: http://firmware.grandstream.com/Release_Note_HT70x_1.0.9.4.pdf I've flashed back to 1.0.6.1 and have it working again, but I would like to try the update again, if I can narrow in on the issue. I'm wondering if anyone can discern the likely problem area from reviewing the change notes? OE

First off, please excuse this probably simple question, but I have seriously tried to wrap my head around it and to find an answer without luck. I am trying to understand the difference(s) between SIP TRUNKS and SIP CREDENTIALS. I mean, there are plenty of VoIP providers out there that provide services to end-users, and provide the credentials for using their service. Are these services basically SIP TRUNKS - or is a trunk actually different?