In your voicemail.conf file use the line below to get asterisk to run esmtp as root even when started by a boot script. Be sure to replace the exact directory location of esmtp/sendmail with yours.

mailcmd=su -c “/opt/bin/esmtp -t -f your@email.com”

You could also replace the command with the sendmail equivalent since esmtp mimics sendmail to a certain degree. The command I listed isnt rocket science, but i just didnt know enough about linux to actually implement it correctly in the voicemail.conf file. Hopefully this helps someone figure out why asterisk will not send emails with the voicemail as an attachment unless run manually via command line and not by a boot script.

All the rage in the VOIP world is the Obihai obi110 voice gateway device, and while due largely to its native integration of google voice, I believe it’s because it does quite a bit for a very reasonable cost. And while just good news for asterisk 1.8 users, this is great news for asterisk 1.4 and 1.6 users! The Obi110 is essentially a user friendly linksys spa-3102 with a slightly watered down interface, with the addition of native google voice support (and setup wizards!). It is somewhat less capable in functionality (no router feature, or independent trunking capabilities with asterisk, etc.) but for my purposes, I decided to give it a try to see if I could replace my spa-3102 and found out that it could quite easily.

The obi110

I have a simple asterisk 1.4 based setup with several extensions located around the world. The main goal is for my family to connect to my system using their iphones, computers, voip devices and make calls to the US and also to receive calls with local DID numbers to their extensions. My previous setup with the linksys spa-3102 allowed them to use my ooma device to call out as an outgoing trunk, however I wanted to see if I could add two trunks to my setup and give my users the flexibility of outgoing call redundancy if they so desired (choose to call out through GV or my Ooma).

Below is my guide for setting this up.

Pre-requisites:
-Network connectivity to the obi110 and ability to administer its local web admin page and setup through obitalk.com.
-Physically connected pstn service (ooma, telco) to the obi110 local Line port (Note: if line port is disconnected, the obi110 will report the lines as busy even if just calling out through GV when the SP2 local dialplan shown below is specified).
-Newest version of firmware on the obi110 (as of 03/21/11 version 1.1.0 (Build: 1892))

Configuration:
I found that it is much simpler to initially set up the obi110 through the obitalk.com portal and then to disable auto-provisioning and make customized changes to the web admin afterwards.

Google Voice Trunk setup – obitalk.com portal
1. Goto obi110 device portal after you have successfully registered your hardware to your user account on obitalk.com
2. Configure Voice services for Service Provider 1 and select Google voice
3. Enter in your google voice email and password.
4. Make sure to put a checkmark to make this provider the default Line to call out.
5. Click submit and wait for your ob110 to reboot

Google Voice Service provider 1 Trunk

Asterisk Trunk setup - obitalk.com portal
1. While still logged into the Obitalk portal, configure the Voice service for Service Provider 2 as a generic service provider (this will essentially act as an extension to asterisk)
2. Enter in your asterisk server information
3. Click submit and wait for the obi110 to reboot.

Asterisk Service provider 2 Trunk

Configure Dialplan on Obi110 device and disable Auto Provisioning
Thats all there is to setting up obi110 trunks! Couldnt be easier right? Now for the small dialplan customizations to make the incoming and outgoing calls route properly.

1. Log into your obi110 device through its web admin (enter in the devices IP address in a web browser. default user/pass is admin/admin)
2. Disable Auto-Provisioning by expanding the System Management Tree on the left pane, then clicking on the Auto Provisioning link and set the Auto Provisioning Method to Disabled. Click submit and reboot (if not disabled, your changes will get overwritten).
3. Expand the Voice Services Tree to the left and click on the SP1 Service link
4. Under X_InboundCallRoute enter in SP2 and click on submit (this routes all calls coming in from GV to your asterisk extension).

Service Provider 1 Local Dialplan

5. Under the Voice Services Tree to the left, click on the SP2 Service link
6. Under X_InboundCallRoute enter in {>(xxx xxx xxxx):sp1},{>(1xxx xxx xxxx):li1}  and click on submit

Service Provider 2 Local Dialplan

The above line in Step 6 routes any 10 digit calls out through GV (configured as SP1), and any 11 digit calls starting with 1 out through the local line port…In my case, through my ooma device). The reason you will need to do this is because unlike the spa-3102, there is no seperate interfacing for asterisk to trunk to the different Service providers which the obi110 can be configured. Instead, asterisk needs to send the dialed number in a way where the obi110 can then determine to which interface it will send its calls. In this case, I will have asterisk format the numbers to be only 10 digits if a user wants to call out through GV, and append a 1 to the 10 digit number if the user wants to call out through my ooma. Once asterisk does this and sends the number to the obi110, the obi110 will then direct the call accordingly depending on if there is a 1 in front of the number – a little hurdle, but easy enough to work around.

Double checking your Asterisk configuration settings
1. Under the Service Providers Tree to the left, click on the ITSP Profile B link
2. Make sure the settings are correct for the SIP and RTP sections. Most likely you will have to modify your RTP LocalPortMin and LocalPortMax settings to match your current asterisk RTP port range.

Thats all there is to configuring your local OBi110 device. The next steps are just to configure asterisk with the user you specified in the obitalk.com service provider 2 settings, and to create a dialplan which handles the numbers so that obi110 can route the calls appropriately.

Below is my user configuration for asterisk 1.4 so that the Obi110 can register to it. ‘gvtrunk’ is my username and ‘password’ is the password which I specified in my Service Provider 2 setup on the obitalk.com portal. Place this in your sip.conf or users.conf file.

[gvtrunk]
username = gvtrunk
fromuser = gvtrunk
secret = password
type = friend
disallow = all
allow = ulaw
allow = alaw
allow = gsm
context = gv-in
host = dynamic

Next we have to edit our extensions.conf file and to tell asterisk what to do with our calls. Below, any call the user makes whether be it 1XXX-XXX-XXXX or XXX-XXX-XXXX will be routed out to the obi110 and then through google voice. This is because I have asterisk removing the 1 digit the user has dialed so that the obi110 dialplan will route the 10 digit number to GV.

In this dialplan I also have any number with a 9 dialed before it sent out to the obi110 as a 1XXX-XXX-XXXX number. If the user has dialed a 9XXX-XXX-XXXX number without the 1, asterisk will append the 1 in front of the number to make it an 11 digit number so that the obi110 can route the call through the local line port (and out through my ooma).

_XXXX is needed because my internal extension numbers are 4 digits long. I need this so that any numbers longer than 4 digits go out appropriately and are not seen as internal extensions.

[ob110-out]
;make outgoing calls to gvtrunk with 10 digit dialing
exten => _XXXX.,1,Dial(SIP/${EXTEN}@gvtrunk,20)
exten => _XXXX.,n,Hangup()
exten => _XXXX.,n,Congestion
;make outgoing calls to gvtrunk with 11 digit dialing
exten => _1XXXX.,1,Dial(SIP/${EXTEN:1}@gvtrunk,20)
exten => _1XXXX.,n,Hangup()
exten => _1XXXX.,n,Congestion
;make outgoing calls to gvtrunk through ooma with 10 digit dialing
exten => _9XXXX.,1,Set(CHEXTEN=${EXTEN:1})
exten => _9XXXX.,n,Dial(SIP/1${CHEXTEN}@gvtrunk,20)
exten => _9XXXX.,n,Hangup()
exten => _9XXXX.,n,Congestion
;make outgoing calls to gvtrunk through ooma with 11 digit dialing
exten => _91XXXX.,1,Dial(SIP/${EXTEN:1}@gvtrunk,20)
exten => _91XXXX.,n,Hangup()
exten => _91XXXX.,n,Congestion

How to handle inbound calls from GV with Asterisk
The last thing we need to do is tell asterisk how to handle inbound calls from Google Voice. In my user setup above, I specified gv-in as the user context. This just tells asterisk where to go and how to handle the incoming calls from the obi110 device. Place this in your extensions.conf file and modify it accordingly.

[gv-in]
exten = s,1,answer
exten = s,n,wait(1)
exten = s,n,Dial(SIP/4000,20)
exten = s,n,wait(1)
exten = s,n,Voicemail(4000,u)
exten = s,n,Hangup

Above, I have all calls from GV calling my extension 4000. Just change it to your desired extension.

Thats all there is to it. It may look complicated, but the dirty work is done for you…Enjoy!

Cisco 79xx IP Phone

In fact, Cisco phones are simple. They just need a tftp server to update its firmware and it doesnt even have to be located on the asterisk/phone system itself. 7940/60 Cisco phones update in two stages. The bootloader, then the application. Once the bootloader is upgraded, all passwords and networking information on the phone will be wiped out. But what is essential in upgrading these phones is a DHCP server with option 150 enabled – an option to auto-provision the tftp server ip address when the DHCP server hands out the regular ip address to the phone.
Without an auto-provisioned tftp server address, the phone will only complete half the upgrade and come up with an error message of “protocol application invalid.” This just means that the phone could not complete the second half of the upgrade and needs the tftp ip address to be specified. Only problem is since there is no application running, there is no place to specify the address! Thank goodness for DHCP option 150 which does this for us when the phone grabs an IP address after loading the bootloader.

Below are instructions to upgrade/convert any Cisco 7940, 7941, 7960, 7961, 7970, 7971, 7942, 7962, 7945, 7965 IP phone to any firmware you want from the Cisco web site (SIP/SCCP), and to recover any cisco phone which may have screwed up somehow while upgrading firmware (like if you get the protocol application invalid message).

Step 1: Download Cisco phone firmware
Cisco may require you to have a smartnet contract before you download the firmware, but if you do not have one, there are firmware files available on the internet. Ive provided two sources below (If you cant find any after looking, for a donation I may be able to get what you need).

a. Goto http://www.cisco.com/ and create a login if you do not already have one. Download the firmware for the specific phone you are using. (if cisco doesnt allow you to download the firmware, check here: ftp://ftp.itl.ua/pub/cisco/ip-7900ser/ or http://radiotwenterand.nl/~graver/cisco/SIP-7960/ for phone firmware).

b. Download the .zip file and extract it to a directory. If you are downloading an older version of the SIP/SCCP firmware and it only comes in a .cop file extension. Just rename the extension to .tar.gz and extract it to a directory using winzip or winrar.

Extracted files in a folder

You will have several files in this directory. If you just have one, you must remember to extract all the files.

Step 2: Download and configure the DHCP/TFTP server
We are going to use a windows based machine and load tftp/dhcp software on it which will make our windows machine a server for our phones.

a. Go here to download the tftp/dhcp server – http://tftpd32.jounin.net/
Download and install version 3.23 (<–YOU MUST DOWNLOAD AND INSTALL THIS VERSION. I have had many people complaining about their phones not obtaining a correct ip address and also the files not being sent correctly. Every one of these problems was due to using the incorrect version and/or having their firewall on).

b. You will also have to configure your windows machine to use ip address of 192.168.1.1 and subnet mask of 255.255.255.0.
Youll configure this tftp server to host the upgrade files via tftp server (theres an option to configure the working directory in the program to use for the stored firmware files on the windows machine – this must be set correctly!).

c. Under DHCP server within the tftp32 program, enter the following and click SAVE:

-IP pool starting address as 192.168.1.2
-Size of pool as 250 (if doing multiple phones, this just makes sure your phones are able to receive an ip address)
-Default router as 192.168.1.1
-Mask as 255.255.255.0
-Additional option, enter 150 in the first box, then 0x0101A8C0  in the second box (this number is 192.168.1.1  in Hexadecimal).

TFTP example

3. Putting it all together

a. Now, there are a few things to consider when upgrading 7940/7960 cisco phones (these steps below dont apply to the 79×1/79×2/79×5 java based phones, if you have one of these, just do steps d. and e.). But the rule of thumb, is that if you have SCCP on your phone youll need to use a XMLDefault.cnf.xml file to specify the firmware load information tag by which the phone can know what load it should install. For the rest of this guide I will just assume you want to convert SCCP to SIP.

b. The easiest way to get SIP on your phone is to install a universal application bootloader. Ive found that SCCP version 8.0.9 (P00308000900) has the most compatible loader which easily allows swapping between current SIP and SCCP loads without a hitch. So the best thing to do is to upgrade your SCCP phone with a newer SCCP load before switching over to SIP. If you have a very old version of SCCP, you will find that you cannot immediately upgrade to this version. If that is the case, just load version 3 (P00303010102), then load to version 5 (P00305000500), then jump to version 8.0.9 (P00308000900). To do this you have to download those firmware versions and just extract them all into one directory.

c. Now download and place this file into your firmware directory. XMLDefault.cnf.xml (right click on link…’save target as’ – this file contains the text below)

<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>
<loadInformation>SCCP40.8-3-1S</loadInformation>
<authenticationURL></authenticationURL>
<directoryURL></directoryURL>
<idleURL></idleURL>
<informationURL></informationURL>
<messagesURL></messagesURL>
<servicesURL></servicesURL>
</Default>

After you have downloaded the file, edit the loadInformation tag within the XMLDefault.cnf.xml file with the firmware you want to install on your phone. Do not add the file name extension, just the name of the file (ig. <loadInformation>P00308000500</loadInformation>). Save and proceed to the next step.

d. Next plug in the phone by crossover cable or ethernet switch to the computer you are running this server from (make sure you disconnect any device attached to the network which resolves an ip through dhcp, as this will conflict with any other dhcp server on the network).

e. Power Cycle the phone and clear the configuration by holding down the # key and then plugging in the phone. After all the lights cycle on the phone, release the # key and press 123456789*0# followed with the 2 key (if you are using a 79×1, 79×2, 79×5, 7970 phone, plug the phone in and hold the # key till the extension lights blink amber…then press 3491672850*#. The lights will turn red, but if they dont, repeat this step. The phone will then wipe the flash and load whatever firmware is specified in the tftpd32 directory – if the phone just keeps downloading the term.defaults file constantly, just do an incremental upgrade to the firmware version that you want).

This will clear all settings on the phone and set it to defaults. Upon reboot, the phone will grab an ip address from the dhcp server on the windows machine running the dhcp server. Tftpd32 will also assign an ip address for the tftp server to the phone using option 150, and the phone should start pulling down files from the server after it obtains an ip address.

f. If you are doing an incremental upgrade to get to version 8.0.9, just keep editing the XMLDefault.cnf.xml file with the next version of firmware and save, then reboot the phone (after it has completed loading both the bootloader and application). The phone should find that it needs to upgrade itself until you stop editing the file with new firmware information.

g. After the phone gets to SCCP version 8.0.9, all you have to do is just download the version of SIP firmware you want, and then extract it into a seperate directory. Point the tftpd32 program to use that directory, copy the XMLDefault.cnf.xml file to that directory and edit the loadInformation tag with the POS-xxx filename which is located in that directory and  reboot the phone. The phone will automatically switch over to loading the SIP firmware.

You can check the events viewer within tftpd32 to ensure that the phone is grabbing the necessary files and getting an ip address. The 79×1/79×2/79×5/7970 phones take a bit longer to fully complete the flashing process since they use more numerous and larger files than the 7940/60 series phones. If the phones just sit like they arent doing anything, wait a bit longer and they should come back up. Its not uncommon for one phone to take up to 4-5 minutes to complete the entire process.

Thats all there is to it, and now that the phone has a good universal application bootloader, you will have no trouble switching between SCCP and SIP by just pointing the phone to the appropriate directory.

If you have any questions, feel free to send me a message on my contact page. If you get really stuck I can also provide these services remotely.

Linksys SPA-942

Asterisk does not easily work with shared line appearances unless the phones are made specifically for it. However, you should be able to get BLF working instead of SLA and have almost the same functionality. Below is what you will need to do to enable BLF on asterisk 1.4 using a linksys spa942 (this should also work on any Linksys/Sipura Phone that supports BLF and extended functions).

Edit your SIP.CONF file and enable these features (these MUST be enabled for hints to work correctly):

allowsubscribe=yes
limitonpeers=yes
notifyringing=yes
notifyhold=yes

In your SIP.CONF/USERS.CONF file, edit your users to contain the following code:

qualify=yes (im not sure this is necessary as I tested it without this setting and BLF still functioned correctly)
call-limit=100 (this can be anything, but 100 will keep you from denying any new calls)

In your EXTENSIONS.CONF file, you just have to enable hints by putting this in your default context (change the extension numbers with your extension numbers):

exten => 1000,hint,SIP/1000
exten => 2000,hint,SIP/2000
exten => 3000,hint,SIP/3000
exten => 4000,hint,SIP/4000

This enables BLF on your asterisk server. In order to configure your linksys spa-942, all you have to do is enter a few settings on the phone itself. But you must be sure you are currently running the newest version of the phones firmware (it is currently at 6.1.5(a)), BLF will not work without first running the newest firmware release! After you have confirmed this, use your web browser and enter the ip adrress of your phone. Click on admin login and then on advanced.

Under the Phone tab type the following:

Short Name: Anything you want to show up next to the LED to remind you which line it is you are monitoring.
Extended Function: fnc=blf+sd+cp;sub=5000@$PROXY;ext=5000@$PROXY 

Just enter the line under extended function

 Above, I have two extensions; 5000 and 6000 which I am monitoring which show up on my phone line buttons 2 and 3. After these settings are entered, scroll down to Line Key Extended Function and enter these settings:

Line Key Extended Functions

Server Type: Asterisk (*this is important!).

You do not have to enable line keys in the Ext. tabs  for this to work. Just click ‘submit all’ and wait for the phone to reboot. Youre done! Hopefully this will save someone lots of time searching google for consolidated information!

While my family uses this setup, I often wondered the details of the calls made out through my Ooma device. Sure I could log into my.ooma.com or even check the asterisk call logs, but it wasnt as neat/tidy or as automated as I would have liked. I snooped around and discovered that Asterisk can run system commands when they are specified in the extensions.conf file!

I scoured the internet for instructions and realized the lack of a good guide for asterisk 1.4 when using sendmail (this guide should work for msmtp as well), so I thought i would share.

Below is my dialplan for sending an email alert when a call is placed through to my ooma device. This can be placed in any dialplan where you would like an email alert to be made to you, so the possibilities are endless!

While many guides are written for use with the linux mail application which can allow for the subject and body to be specified by the command line, I was using sendmail which does not allow for the subject or body to be specified by command line (except using echo statements in a way which asterisk could not duplicate while running from a dialplan ig. (cat > text | echo xxx ; echo xxx ; | sendmail) etc.). Sendmail needs certain variables to be specified in a file, or by running the application and specifying it while it is running.

I found that asterisk would need to create the file dynamically and add the To:, Subject:, and Body text variables before Asterisk/Sendmail would send the customized email that I wanted sent.

*Note: For this to work, you must have sendmail already configured and able to send emails.

[ooma-out]
;make outgoing calls to ooma pstn
exten => _XXXX.,1,System(echo “To: putdestinationemail@here.com” > /opt/etc/init.d/calls)
exten => _XXXX.,n(done),NoOp()
exten => _XXXX.,n,System(echo “Subject: [PBX]: Outgoing call through Ooma” >> /opt/etc/init.d/calls)
exten => _XXXX.,n(done),NoOp()
exten => _XXXX.,n,System(echo “” >> /opt/etc/init.d/calls)
exten => _XXXX.,n(done),NoOp()
exten => _XXXX.,n,System(echo “User ${CALLERID(NUM)} has made an outgoing call through ooma to phone number ${EXTEN} on ${STRFTIME(%C%m%d%y%H%M)}” >> /opt/etc/init.d/calls)
exten => _XXXX.,n(done),NoOp()
exten => _XXXX.,n,System(sendmail -t -f putsendingemail@here.com < /opt/etc/init.d/calls)
exten => _XXXX.,n(done),NoOp()
exten => _XXXX.,n,Dial(SIP/${EXTEN}@pstn,20)
exten => _XXXX.,n,Hangup
exten => _XXXX.,n,Congestion

exten => _XXXX., is used instead of exten => s,n, because I wanted to have any number larger than 4 digits routed out through ooma, since my internal extensions are 4 digits long.
exten => _XXXX.,n,Dial(SIP/${EXTEN}@pstn,20) is my command to send the call out through ooma.

This configuration makes a file located in /opt/etc/init.d called “calls” containing:

To: putdestinationemail@here.com
Subject: [PBX]: Outgoing call through Ooma
User EXTENTIONCALLING has made an outgoing call through ooma to phone number NUMBERCALLED on DATE

The email is sent to the email specified in “putdestinationemail@here.com” from email address specified in putsendingemail@here.com

This dialplan can be customized to suit any email alert you may need to send when placed into any dialplan in the extensions.conf file, so have fun!

I freaked out for a moment as I believed there was no way to recover ssh access…until I thought about it some more. Below are the steps to recover SSH access to your pogoplug running openpogo if you somehow disabled hbmgr.sh on startup:

This procedure will work with pogoplugs running openpogo, not sure if it will work for units NOT running openpogo since openpogo searches the opt/etc/init.d directory on startup to start any other scripts on boot.

PREP:
-power off the pogoplug
-download a linux live distro. Any modern live distro will do, knoppix, slitaz, ubuntu, etc.
-burn distro to cd
-boot off linux live cd
-plug in HD from pogoplug into the computer now running linux

ACCESS the HD:

-mount hard drive (you might have to open up gparted to see what device your HD is coming up as)
-start terminal
-access your hd mount
-do an ls -a (this command shows hidden files on the directory you are in. If you do not use the -a command, you will only see the directories you shared out through pogoplug).
-go into the /.opt/etc/init.d directory
-nano or vi a new file called S70ssh with ROOT permissions (otherwise you will not be able to write the file)

Enter in this line in the new file:

/etc/init.d/hbmgr.sh start

-save and exit
-chmod 777 the S70ssh file for good measure

FINISHING:
-Power down the linux system
-Plug the HD back into the Pogoplug and power cycle the unit.

After the unit restarts, the script should have run and started hbmgr.sh. Remember to delete the script S70ssh and make sure the /etc/init.d/hbmgr.sh start line is enabled or entered back into the rcS startup file.

After some searching, I found that plugbox linux changes up the boot parameters using blparam. I figured that the /sbin/reboot was indeed working fine, but it would just freak out when trying to check for installed linux kernels on attached USB storage devices upon reboot. Reverting to the original bootload parameter worked for me and fixed /sbin/reboot so I didnt have to power cycle the unit by physically pulling the plug.

I didnt find this command easily by searching how to fix a broken reboot command so I figured I would post it on my blog for others who may run into the same problem.

To fix /sbin/reboot, SSH into your device, enable rw access to the file system and enter in:

/usr/local/cloudengines/bin/./blparam ‘bootcmd=run bootcmd_original’

you should be able to issue an /sbin/reboot and have it reboot like normal!

Seagate Dockstar

After setting up the dockstar, I installed openpogo as I wanted it to replace my NSLU2 which was acting as a transmission bit-torrent downloader and Asterisk PBX. It all installed pretty much ok, but with the exception of setting a static IP for the unit. I initially just set the DHCP options in my router to give it a static IP, but I wanted to know if there was an easy way to do this with the pogoplug software. I ended up emailing pogo support and below is what they told me:

Please read through these instructions carefully. Applying them improperly will result in a bricked or inoperable Pogoplug. By the way, we already have logged an enhancement request to add this functionality in the Web interface at my.pogoplug.com though I have no information regarding a release date for such an update.

You will need to first activate Pogoplug using the default setup I.S.P. modem => router => Pogoplug with an functioning DHCP Server.

Next enable SSH: To enable SSH, login to http://my.pogoplug.com, navigate to the Settings page, select Security Settings, and from the main pane, enable SSH for your Pogoplug. Choose your own SSH password.

Now SSH into your Pogoplug with the username: root and the password you set previously.

At the bash shell prompt, you can assign a static IP just as you would on any linux machine. For this example, I will use eth0:3 aliased interface so we don’t lose our current static IP we are using for SSH. I will also assume my new static IP to be assigned is 192.168.77.3 and the default route gateway is 192.168.77.1 and the DNS name server is 192.168.77.2

To persist these changes after a power loss or reboot of the pogoplug, issue the following commands at the bash prompt.
-bash-3.2# mount / -o remount,rw,noatime
-bash-3.2# echo ‘ifconfig eth0:3 192.168.77.3 netmask 255.255.255.0′ >> /etc/init.d/rcS
-bash-3.2# echo ‘route add default gw 192.168.77.1′ >> /etc/init.d/rcS
-bash-3.2# echo ‘echo “nameserver 192.168.77.2″ > /etc/resolv.conf’ >> /etc/init.d/rcS
-bash-3.2# mount / -o remount,ro
-bash-3.2#

When done properly, your Pogoplug should continue to function when attached to a DHCP network.

I wanted to post this because I could not easily find it online or in their forums, so I thought this could save someone the time and hassle of contacting their technical support.

The real motivation for me to use asterisk and ooma was to save moolah. With ooma, albeit with the up-front cost of $200, one can cancel their phone line and stop paying those pesky monthly bills. Our monthly bills were not as absorbitant as others, yet the reasoning behind me getting it was to get more for the same. I canceled my ATT $10 internet (768k down) and $10 phone with only local calling…(total monthly charge of $32-$37 with taxes and long distance charges that we didnt make), and opted for a 12mb down 512k up cable internet connection for $37.99. I would then still have a home phone number which I could take with me if I moved, and have super fast internet (in my standards). With asterisk in the mix, I could then share out my ooma with my family and allow them to make calls to the US for free as well (my sister lives in australia). Also, if I am overseas, I can make free calls to the US. I also have gizmo5 and google voice working together to provide my individual extensions in asterisk with DID numbers, but thats for another post.

For those of you who do not know what ooma is, its a VOIP hardware solution which gives you a dedicated phone line and “unlimited” calling to the US, all you need is an internet connection. I got my ooma core from radio shack for $199 and will never have to pay phone bills again (ooma core does not have an annual regulatory fee, while the ooma telo, and ooma hub only, does). Ooma makes its money off the upfront cost of hardware, and also by selling its ooma premier service. This service gives you cool features, but features that asterisk allows you to do and with more customization (and maybe for a little more effort). The Linksys SPA-3102 is an ethernet voice gateway with FXO port that has the added functionality of routing, and it also acts as an ATA to allow your regular analog phone to connect with a VOIP provider using the FXS port.

Ooma hub wiring setup

The Ooma hub can be hooked up to your existing phone lines in several ways. Currently, I have my ooma phone port plugged directly into my existing home wiring jack with a splitter which also has my fax/answering machine plugged into it. This configuration allows all the phone jacks in my home to access the ooma hub without the use of the ooma scout. This is essentially the same wiring configuration as one receiving phone service from the telco. However, you lose the instant second line feature you would otherwise be provided when using the ooma scout adapter. To connect the SPA-3102, just plug a phone cord from the Line port on the adapter to a jack in the wall, or if it is near the ooma hub, into a splitter which shares a line plugged into the Phone port on the Ooma hub (or directly into the hubs phone port without a splitter). If you are on a call using a phone plugged into my wiring configuration and dial out using asterisk through the Line port on the SPA, the adapter will report a 503 message to asterisk and stop the call from taking place and interrupting.

If one wants to make sure the line is not busy when receiving/making a call when using the ooma as a regular analog telephone line as well, another configuration one can use, is to hook up the ooma scout and connect it directly to the ooma hubs ’wall’ port via phone line. One would then connect a phone line from the SPA-3102 Line port to the scouts ’wall’ port. This enables the scout to communicate with the ooma hub and enables the instant second line feature should the first line be active when a call out from the asterisk box takes place.

Linksys SPA-3102 Configuration

SPA-3102 Remote Management for LAN Setup

The SPA-3102 has four interface ports in the rear, Internet, Ethernet, Phone, and Line. If you plug in a computer to the ethernet port via cable, it will provide your computer with an ip address with which you can then enter in the gateway address from an ipconfig and hit the spa3102 web gui. With this web gui, youll be able to configure the device. We dont want to have to plug in a cable each time to configure the device, so we will enable the web interface on the spa3102 when it is connected to the Internet port (with which it will receive a dhcp address handed out to it from your router currently on your network).

-Log into the webgui for the spa-3102 when you are connected to the ethernet port
-Click on the admin and advanced links at the top right to get the elevated setup access
-Goto ‘WAN Setup’ Tab
-Change ‘remote management’ option to ‘yes’
-Click the ‘submit all changes’ button at the bottom.

-Connect your spa-3102 to your network via the internet port.
-Log into your spa-3102 and look at the status screen with the computer still plugged into the ethernet port on spa3102. You will find the dhcp address the spa-3102 received from your router which is connected to the Internet port.
-Disconnect your computer from the ethernet port on the spa-3102
-Log into your 3102 via the dhcp address that it received from the internet port.

SPA-3102 PSTN Line Setup:

Now we begin the configuration of the SPA to be used with asterisk. In this setup, I will not enable Line1 which makes the SPA-3102 an ATA adapter as well (allowing calls made to your voip provider to ring the analog phone connected to the PHONE port). In this setup, I just use the SPA as a gateway which allows me to make and receive calls (using the LINE port on the SPA) from any extension that is connected to my asterisk pbx. Under the LINE 1 tab in the SPA, ive set ”Line enable” to no.

NOTE: When things are configured properly, and the PSTN Line is registering with asterisk, the LINE LED on the SPA will light up and remain lit (same with the Phone port if Line 1 is enabled). If things arent communicating correctly, the LED will not be lit (I have the spa registering to asterisk).

-Log into the SPA web gui
-Click on the admin and advanced links at the top right to get the elevated setup access
-Click the ‘PSTN Line’ tab

Where you enter in your asterisk server IP info and whether or not it will register to asterisk.

Proxy and Registration

-Proxy – Change 192.168.1.77 to the ip address asterisk is on your network. I put the ip in the outbound proxy, its not necessary as ‘Use Outbound Proxy’ is set to ‘no.’
-Register – ‘yes’
-Make call without reg and Ans call without reg – Change options to ‘no’

Subscriber Information

-Display name – can be anything
-User Id – can be anything but for simplicity sakes when configuring asterisk, use a name without spaces
-Password - can be anything

This is where you enter dialplan information, and options to configure sending calls from asterisk to the pstn port.

Dial Plans

This section of dial plans are accessed by the entire page of the SPA PSTN Line tab. There are eight DP fields because it allows you to create different dial plan options to be used throughout this tab. Voip-to-PSTN, and PSTN-to-VOIP sections both reference these dialplan fields as ‘DP.’ As you can see in my screenshot, Dial Plan 2: is filled out. In my setup, this command tells the SPA that any calls answered after the PSTN-to-VOIP gateway option answer delay is reached, to be sent to the S extension in asterisk. You may enter any extension such as (S0<:102@asteriskIP>).

VOIP-to-PSTN Gateway Setup

-VOIP-to-PSTN gateway enable – ‘yes’
-Line caller DP – set to ’1′ (this option references Dial Plan 1:  and the default (xx.). This just passes anything sent from asterisk to the SPA without any change)
-One Stage Dialing – set to ‘yes.’ If set to no, then the SPA uses 2 stage dialing, and it screws up asterisks calling out to the Line port.

The PSTN-to-VOIP configuration, where you configure how to send calls from the Line port (pstn) to asterisk

PSTN-to-VOIP Gateway Setup

-PSTN-to-VOIP Gateway enable - ’yes’ (in the screenshot above, i have it set to ‘no’ as I do not want the spa to pick up the line and forward to my extension s@192.168.1.77 as defined in Dial Plan 2:. This essentially turns off any calls going to my asterisk system. I have my asterisk system setup to forward the call from the pstn to my cell phone when this is turned on, and only used while traveling far away from home. When I am not traveling I have a fax/answering machine on my ooma and want it to pick up instead, so it is disabled.

If you are in an asterisk/voip only configuration and want all calls to be routed straight to your asterisk system without worrying about any analog answering machines or fax picking up/ringing, then set to ‘yes’

-PSTN CID For VoIP CID – set to ‘yes’ if you want callerid to be passed onto your asterisk system
-PSTN Caller Default DP – set to ’2′ as in my Dial Plan 2: it allows calls to be routed from Line port (pstn), to extension S on my asterisk pbx

FXO timer values (sec)

-PSTN answer delay - this option is to change the length of time the SPA-3102 picks up the call coming in from the PSTN and forwards it to your asterisk system. The default is 16 which allows the line to ring for a little too long before sending it off to Asterisk. A number of 3-5 should be good for callerid to be gathered and sent along with the call to asterisk.
-PSTN Dial Digit Len – set to .1/.1 otherwise calls may take longer to start connecting. This essentially shortens the speed at which digits are dialed at. You dont want digits to take forever to be entered do you?

Awesome. You are now finished configuring your SPA-3102 to act as a SIP trunk on the SPA-3102 side. All you need from here to configure asterisk is the username and password you specified in the Proxy and Registration section. Now on to the Asterisk side of things…

Asterisk Configuration

This setup is for asterisk 1.4. I found that many guides found on the internet do not seem to work for my setup. I was not able to set my SPA as a peer, but had to configure my 3102 to register to asterisk as an extension in order for everything to work correctly. I also found many internet guides had sip trunk settings which were no longer used for version 1.4. Now lets tell asterisk theres a device to communicate with in the Users.conf (for you it might be Sip.conf) file in the asterisk directory.

Users.conf or Sip.conf

Users.conf or sip.conf configuration in asterisk. this command sets up the SPA to be used as an extension from which calls can be made and received.

-[pstn] – Put the username you specified on the SPA-3102 in between brackets. In my example above, replace [pstn] with your username. The tricky part here is that the name between brackets is your username, even if you specify username = as something else, asterisk will not allow your 3102 to register with it.
-type = friend - Sets as an extension which can be dialed out from (see here for more info)
-port – I saw several guides saying to put in port = 5061, this is unecessary as this configuration automatically connects to this port. You may need this if it was set to type=peer, but I was never able to get it to work as one.
-disallow – This configuration also disallows any voice codec other than ulaw, alaw, and gsm as the NSLU2 does not have enough horsepower to transcode the higher compression the other codecs use.
-context = pstn-in – this is the label for where incoming calls are sent to in my extensions.conf file.
-host = dynamic – as the spa is registering to asterisk, the ip address of the 3102 does not need to be specified here and will be obtained during the spa-3102 device registration. I could change the ip address of the spa-3102 on the unit and it would still register with asterisk.
-secret = passwd - replace ‘passwd’ with the password you entered in the spa-3102

Other than that, this is all thats needed for the PSTN Line to register to asterisk and send and receive calls.

Extensions.conf

Extensions.conf is the the file which tells asterisk how to handle incoming/outgoing phone calls.
Lets configure outgoing calls first, as it requires very little configuration. Below is all i need to put into my default context in order to make outgoing calls out of the Line port (we previously specified ‘pstn’ as a trunk/user in users.conf/sip.conf) on my SPA-3102. Below, the _XXXX. tells asterisk that any number dialed with more than 4 digits should go out through the SPA (I have 4 digit extension numbers configured). Without this configuration, asterisk would try to place a call to an internal extension number, only to find that the extension (the phone number dialed) did not exist.

Simple outgoing dialplan used in the default context in my extensions.conf

In my users.conf (maybe your sip.conf) file, I had the pstn user (pstn is the username specified in the spa3102) use context pstn-in as the label to goto which contains the code for how to handle the calls coming in from the Line port or PSTN.

Extensions.conf tells asterisk how to handle a call. pstn-in is the context defined in my users.conf/sip.conf

In my spa-3102, my Dial Plan 2: had code (S0<:s@192.168.1.77>). This essentially sent the incoming call from the PSTN Line to the S extension in asterisk. Asterisk knows that the call is coming from the pstn user defined in the users.conf/sip.conf file, and found that context pstn-in was specified. It then initiated the commands under the pstn-in section. The commands listed above, answers the call, plays a sound file that nobody is available to take the call, then says “call-forwarding” and proceeds to ring my cell phone number (8001112222 is the cell number, change it as desired. proxy01.sipphone.com is my gizmo account which forwards it out through a google voice DID number).

Alternatively, if you just want to have the incoming pstn call sent to several or all extensions on your asterisk system, you can substitute my ‘Dial’ line with the one below and change my extension numbers to match yours (where 6000, 6100, 6200 are my extensions/phones registered with asterisk):

exten = s,n,Dial(SIP/6000&SIP/6100&SIP/6200,20)

Thats all there is to it, good luck!

New LG VX5500

Software. I know I needed to get some sort of software to interface the phone to the computer, and after some googling, I found Bitpim (www.bitpim.org). Bitpim is a FREE and open source software (free was the magic word!) software that does the same thing (although less elegantly) that other products that cost about $30 do. It supports many phones(*1), and is continually upgraded. Did I also mention its FREE? After one downloads and installs Bitpim, all one needs is the interface to communicate with the phone and in my case, this was either a USB cable or bluetooth adapter.

The teeny tiny bluetooth adapter from Ebay

Hardware. I realized that my laptop did not come equipped with a Bluetooth adapter and buying a USB cable specifically for this phone didnt make much sense as I wouldnt want to have to purchase another cable in the future for any new phone I upgraded to later. I quickly found a good option from ebay. Its a super tiny bluetooth module supported by Windows Vista. It cost less than $5 with free shipping, and arrived in ten days (from Hong Kong). Windows picked it right up and I was now bluetooth enabled. So next thing I needed to do was to get Bitpim to work with the LG Vx5500…heres what I did:

1. Pair your phone with the computer using bluetooth (you will have to put your phone in discovery mode, and follow the prompts given to you by windows after clicking ‘add’ in the bluetooth properties window). After this is done, click on the COM ports tab and view the com port your phone is set to.

My phone is the VX5500, and is connected to COM16

2. Install, download, and run Bitpim (Bitpim does not currently support the VX5500 natively, but by choosing a phone that has been supported some of the features like browsing the filesystem work).

3. Click on the icon with the screwdriver and wrench (settings button).

Bitpims settings window

• Select LG-VX8560 (chocolate 3) <—important.
• Then click on the browse button located next to the COM Port section (in my example, im using COM16, the port my phone is connected to).
• After the window refreshes, select the COM port which was associated to your phone in the previous bluetooth settings window. Click ok and ok to exit out of the settings window.

**NOTE: Bitpim will not autodetect your phone. But by manually specifying the settings listed above, Bitpim will connect and communicate with your phone.**

4. Now you must use the filesystem view/pane to reliably transfer your photos and ringtones. To enable this view, click on the View menu at the top of the window, and put a checkmark next to the View Filesystem option. You should now be able to see the Filesystem Pane/Window.

After enabling filesystem view, and expanding the directory tree

5. It takes a few seconds to get the directory listing after expanding the / folder. Your ringtones will be placed in the 10889/Ringtones folder shown above, and your pictures/wallpapers will be placed in the 10888 folder listed below.

wallpaper/photo directory

6.  Now that you have the locations for your ringtones and wallpapers, here are the guidelines to put them onto the phone.

• Ringtones can be in .mp3 format, but NEED to be renamed to xxx.aac (song.mp3 –> song.aac), then added by right clicking on the pane to the right of the filesystem tree, and clicking on New File… Select your .aac file and upload it. After uploading your files, you must reboot the phone for the file to be seen by the phone. Either power cycle the phone, or right click on the viewing pane and click ‘reboot phone.’

*NOTE* To make the ringtones, I didnt bother with any sound editing software, and just went to http://makeownringtone.com/ and uploaded an mp3. I then used its tools to shorten the song length, and encode it into a lower bitrate, and have it saved as an .aac file. Easy as pie. highly recommend it! Oh yes, try to keep the ringtone to less than 30 seconds…

• Main wallpapers for the LG VX5500 use a pixel dimension of 176 x 220, and front lcd image dimensions are 96 x 64 pixels. Upload these images in the same manner as uploading ringtones to your phone, but using the 10888 directory (you might even see other photos you have taken listed in that directory). Reboot the phone to have the changes take place on the phone itself.

*NOTE* To make the wallpaper, I didnt fuss with any image software. I just took a picture and uploaded it to http://mobopic.com/upload/eng. It allows you to zoom, crop, and just enter in the size of the wallpaper and it does it all for you. Fast, easy, and FREE!

So far, the only thing I want to be able to do now, is to save my phone contact list to my computer. I wasnt able to figure out how to do that, but im sure bitpim will support it in the future. Until then, im not too worried about that as im doing everything that I wanted to do, without the Verizon wireless charge. Once again, I showed my wife, and she made me put ringtones on her phone! Im a hero!

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*1 Phones bitpim currently supports: LG-AX8600, LG-C2000, LG-G4015, LG-LX570 (Musiq), LG-PM225, LG-VX10000 (Voyager), LG-VX3200, LG-VX4400, LG-VX4500, LG-VX4650, LG-VX5200, LG-VX5300, LG-VX6000, LG-VX6100, LG-VX7000, LG-VX8000, LG-VX8100, LG-VX8300, LG-VX8500 (Chocolate), LG-VX8560 (Chocolate 3), LG-VX8600, LG-VX8610 (Decoy), LG-VX8700, LG-VX8800 (Venus), LG-VX9100 (enV 2), LG-VX9700 (Dare), LG-VX9800, LG-VX9900 (enV), E815, E815m, K1m, V325, V325M, V3c, V3cm, V3m, V3mM, V710, V710m, SCH-A870, SCH-A930, SCH-A950, SCH-U470, SCH-U740, SPH-M300MEDIA, SPH-M300PIM, VM4050, LG-UX5000, SPH-A460, SPH-A620 (VGA1000), SPH-A660 (VI660), SPH-A680, SPH-A740, SPH-A840 (Telus), SPH-A840, SPH-A900, SPH-N400, SCP-6600 (Katana), SCP-6650 (Katana-II), SCP-7050, SCP-8400, MM-5600, MM-7400, MM-7500, MM-8300, PM-8200, RL-4920, RL-4930, SCP-200, SCP-2400, SCP-3100, SCP-3200, SCP-4900, SCP-5300, SCP-5400, SCP-5500, SCP-7200, SCP-7300, SCP-8100 (Bell), SCP-8100, VI-2300, LG-LG6190, LG-LG6200, LG-LG8100, LG-LX5450, LG-LX5550, LG-PM325, LG-TM520, LG-VI125, LG-VI5225, LG-V 10, LG-VX4600, LG-VX5400, LG-VX8350, LG-VX8550 (Chocolate 2), LG-VX9400, SCH-A310, SCH-A650, SCH-A670, SK6100, SPH-N200.

I learned much of this information from searching through forums located at www.howardforums.com/. Unfortunately, its scattered and sometimes difficult to understand for a newbie. Hence the blog post.