Eric's Blog

1 Extension, Multiple Phones

eric — Tue, 15 Jan 2008 11:00:21 +0000

In order to setup Asterisk to ring multiple phones from the same dialed extension, you will need to create a phantom extension. I accomplished this by doing the following…

Before we go any further, let’s use the following information as true. The extension we want to have ring in multiple places is extension 100. For sanity’s sake, let’s say we want it to ring in 3 places (regardless of the reason). This means that each phone will need its own extension and auth information according to the sip.conf.

First you need to assign each device (phone), its own extension. Let’s give each device the extension of . Therefore our 3 phones will have the extensions of 1001,1002,1003 respectively. Their entries in the sip.conf will look like this:

[1001]
type=peer
context=internal
username=1001
callerid=Eric Lubow <100>
host=dynamic
auth=1001@192.168.1.2
call-limit=100
nat=no
canreinvite=yes
mailbox=100@allstaff
disallow=all
allow=gsm
allow=ulaw
astdb=chan2ext/SIP/1001=1001

[1002]
type=peer
context=internal
username=1002
callerid=Eric Lubow <100>
host=dynamic
auth=1002@192.168.1.2
call-limit=100
nat=no
canreinvite=yes
mailbox=100@allstaff
disallow=all
allow=gsm
allow=ulaw
astdb=chan2ext/SIP/1002=1002

[1003]
type=peer
context=internal
username=1003
callerid=Eric Lubow <100>
host=dynamic
auth=1003@192.168.1.2
call-limit=100
nat=no
canreinvite=yes
mailbox=100@allstaff
disallow=all
allow=gsm
allow=ulaw
astdb=chan2ext/SIP/1003=1003

Next, in your extensions.conf, add the entry to ring all the extensions when the phantom extension is dialed. The Dial() command should now look as follows:

exten => 100,1,Dial(SIP/1001&SIP/1002&SIP1003,18)

A nice thing to do to (in order to not confuse the user) is, in your tftp files, ensure that the label on the phone (each phone) is still the actual extension of the phone that one would dial to get to it. Label the phone elsewhere with your REAL extension to keep track of it.

Creating a Process Table hash in Perl

eric — Fri, 28 Dec 2007 11:00:01 +0000

I came across a situation where I needed to access the process table in Perl. The problem that i found was that the best accessor Proc::ProcessTable only retrieved an array. Since it seems fairly senseless to keep looping over an array to find the exact process id that I want, you may want to turn it into a hash.

use strict;
use warnings;
use Proc::ProcessTable;

 # Create a new process table object
 my ($pt) = new Proc::ProcessTable;

 # Initialize your process table hash
 my (%pt_hash);

 # Get the fields that your architecture supports
 my (@fields) = $pt->fields;

 # Outer loop for each process id
 foreach my $proc ( @{$pt->table} ) {
    # Inner loop for each field within the process id
    for my $field (@fields) {
       # Add the field to the hash
       $pt_hash{$proc->pid}{$field} = $proc->$field();
    }
 }

It’s just as simple as that. If you want to be sure that its in there. At the end of the file add these two lines for proof:

use Data::Dumper;
print Dumper \%pt_hash;

The hash is organized with the keys being the process ids. There is another hash underneath it with all the fields as hash keys.

MySQL Proxy Query Profiling

eric — Fri, 07 Dec 2007 23:12:44 +0000

Since I am now finally getting to play with MySQL Proxy, I am going to outline some recipes here that I have found/created/modified that may be useful to someone other than me. This is a recipe for profiling queries. It writes the information to the PROXY_LOG_FILE currently name mysql.log. It is a file that will be created in the directory that you run mysql-proxy from. The file itself is mostly commented and should therefore be pretty self-explanatory. It was adapted from the reference documentation listed at the bottom of this entry.

assert(proxy.PROXY_VERSION >= 0x00600,
  "You need to be running mysql-proxy 0.6.0 or greater")

 -- Set up the log file
 local log_file = os.getenv("PROXY_LOG_FILE")
 if (log_file == nil) then
    log_file = "mysql.log"
 end

 -- Open up our log file in append mode
 local fh = io.open(log_file, "a+")

 -- Set some variables
 local original_query = ""
 local executed_query = ""
 local replace = false
 local comment = ""

 -- For profilign
 local profile = 0

-- Push the query onto the Queue
function read_query( packet )
  if string.byte(packet) == proxy.COM_QUERY then
    query = string.sub(packet, 2)

    -- Pull out the comment and store it
    original_query = string.gsub(query, "%s*%*%*%*%s(.+)%s%*%*%*%s*",'')
    comment = string.match(query, "%s*%*%*%*%s(.+)%s%*%*%*%s*")

    -- Add the original packet to the query if we have a comment
    if (comment) then
        if string.match(string.upper(comment), '%s*PROFILE') then
          -- Profile types:
          --   ALL, BLOCK IO, CONTEXT SWITCHES, IPC,
          --        MEMORY, PAGE FAULTS, SOURCE, SWAPS

          -- Set it up for profiling
          -- Generate a new COM_QUERY packet
          --   and inject it with a new id (11)
          original_query = "SET PROFILING=1; " .. original_query
      end -- End if (PROFILE)
    end -- End if (comment)

    executed_query = original_query
    -- Check for a 'SELECT' typo
    if string.match(string.upper(original_query), '^%s*SLECT') then
        executed_query = string.gsub(original_query,'^%s*%w+', 'SELECT')
        replace = true;
    -- matches "CD" as first word of the query
    elseif string.match(string.upper(original_query), '^%s*CD') then
        executed_query = string.gsub(original_query,'^%s*%w+', 'USE')
        replace = true
    end

    -- Postpend the other profiling strings to the query
    if (comment and string.match(string.upper(comment), '%s*PROFILE')) then
        executed_query = executed_query .. "; SHOW " .. comment
    end

    -- Actually execute the query here
    proxy.queries:append(1, string.char(proxy.COM_QUERY) .. executed_query )
    return proxy.PROXY_SEND_QUERY
  else
    executed_query = ""
  end
end

function read_query_result (inj)
  local res = assert(inj.resultset)
  local num_cols = string.byte(res.raw, 1)
  if num_cols > 0 and num_cols < 255 then
     for row in inj.resultset.rows do
         row_count = row_count + 1
     end
  end

  -- Prepend the error tag in the log
  local error_status = ""
  if res.query_status and (res.query_status < 0) then
     error_status = "[ERROR]"
  end

  -- Gets the rows affected by the actual query
  local row_count = 0
  if (res.affected_rows) then
     row_count = res.affected_rows
  end

  -- Prepend the comment line in the log
  if (comment) then
     fh:write( string.format("%s %6d -- [COMMENT] %s\n",
        os.date('%Y-%m-%d %H:%M:%S'),
        proxy.connection.server["thread_id"],
        comment))
  end

  -- Prepend the typo in the log
  if (replace) then
     fh:write( string.format("%s %6d -- [REPLACEMENT] %s\n\t\t\t%s\n",
        os.date('%Y-%m-%d %H:%M:%S'),
        proxy.connection.server["thread_id"],
        ("replaced " .. original_query),
        ("with " .. executed_query)))
  end

  -- Write the query adding the number of rows retrieved and query time
  fh:write( string.format("%s %6d -- %s %s {%d rows} {%d ms}\n",
     os.date('%Y-%m-%d %H:%M:%S'),
     proxy.connection.server["thread_id"],
     error_status,
     executed_query,
     row_count,
     inj.query_time)
  )
  fh:flush()
end

To make this work, simply append 3 asterisks ‘PROFILE ’ and then 3 more asterisks and you will have the profile information returned to you in your query:
*** PROFILE ALL *** SELECT * FROM foo_bar;
2 tables will be returned. Your results and then the profile of your results.

Reference: Writing LUA Scripts for MySQL Proxy

Deleting Lots Of Files (SysAdmin problem solving exercise)

eric — Mon, 17 Dec 2007 11:15:12 +0000

Since I know I am not the first (or the last) to make a typo in logrotate and not catch it for a while…someone else must have been in the position of having to delete a lot of files in the same manner. I recently learned that, as usual, there is more than one way to handle it.

To put the situation in context, I basically allowed thousands of mail.* files to be created. These files littered the /var/log/ directory and basically slowed down the entire file system access. I figured out this a number of ways.

The first way was when I tried to do an ls anywhere, it would just hang. My first reaction was to check to see what was eating up the CPU. To do this, I did a top. I noticed that logrotate was hogging all the CPU cycles. Since I know that logrotate basically only operates on one parent directory (by default) /var/log, I headed on over there and did an ls. Once again, it just hung. Then I figured the file system was slow and decided to check out some file system information. The next two commands I ran were df -h and df -i. I ran the df -h to see if we were out of disk space (and yes I lazily use human readable format). I ran the second to check to see how many inodes were in use. (For more information on inodes, check out the wikipedia entry here).

Now that I know the system is short on inodes, I checked out the output of lsof. Now I know that we have some serious problems in the /var/log dir. After some quick investigation, I realized that there were too many mail.* files. How do I get rid of them? Glad you asked… Let’s assume that we want to delete ALL the mail.* files in the /var/log directory.

1) The easiest way is to do it with find:
1a) Using find’s delete command:
[root@eric] /var/log # find ./ -type f -name "mail.*" -delete
or
1b) using find’s exec command with rm:
[root@eric] /var/log # find ./ -type f -name "mail.*" -exec rm -rf '{}' \;
These will work, but either will be slow since they doesn’t do batch execution.

2) A slightly more preferred way is to use bash:
[root@eric] /var/log # for n in mail.*; do rm -v $n; done;
This is a little faster, but will still be relatively slow since there is no batch execution. (Note: The -v in the rm will cause quite a bit of output since it is showing you EVERY file it deletes. Feel free to leave this out if you really screwed up.)

3) The actual preferred method is to use find:
[root@eric] /var/log # find ./ -type f -name "mail.*" | xargs rm -f
I believe this is the preferred method because although it removes the files one at a time, it is more efficient for the file system since it batches it up.

There are certainly other ways to accomplish this task. It can always be done with a Perl one-liner or even using some Perl modules to save some time. These are just a few ideas to point someone in the right direction.

Text Messages to Cell Phones via Email

eric — Mon, 03 Dec 2007 19:23:17 +0000

I have been compiling a list of the domains that one needs in order to send text messages to cell phones via email. As a huge user of Nagios, this is how I keep myself aware of the status changes. Below I have listed the carriers that I use most frequently. If you have any others to list here to make this more complete, please add a comment and I will add it to the list.

I have seen other instances of this before, but some are outdated. These are the newest ones that I have come across.

The assumption here is that the telephone number of the person that you are trying to text message is 2225551212. Just make sure that there is nothing in between the numbers (like a ‘.’ or a ‘-’), Also make sure that you don’t put the ‘1′ before the phone number.

ATT: 2225551212@txt.att.net
Verizon: 2225551212@vtext.com
T-Mobile: 2225551212@tmomail.net
Alltell: 2225551212@message.alltel.com
Virgin Mobile:2225551212@vmobl.com
Sprint: 2225551212@messaging.sprintpcs.com
Nextel: 2225551212@messaging.nextel.com
All Other: 2225551212@teleflip.com

It should be noted that the last item (Teleflip), can be used either in place of any of these or as a fall through. It seems to act as a universal text message system.

UPDATES:
Here are the contributed addresses. The thanks are in parentheses following the numbers:

Claro (Brazil): 2225551212@clarotorpedo.com.br (Rodrigo)

Cloning a Virtual Machine in VMWare VI3 without Virtual Server

eric — Mon, 05 Nov 2007 18:26:24 +0000

I, like many other people working in a small company, have to fix problems and come up with solutions with cost at the forefront. I had to make many virtual machines appear from nowhere to create an environment in virtually no time at all. Since all I had was VMWare Server (for Linux), I started there. When I realized that those didn’t translate to ESX, I had to come up with another solution. I created a single template guest OS (of Gentoo 2006.1 which is our primary server OS here) and decided to clone that. How did I do it…well, I am glad you asked.

The key here was to figure out what the VI3 (Virtual Infrastructure 3) client did and mimic it. In order to figure this out, I copied the entire /etc directory to a place where I could later diff it. I created 3 VM (virtual machines) with nothing on them to discern the patterns that the client made in its files. I then diff’d the 2 version of the /etc directory and now I knew the main changes that had to be made. It also should be noted that the Temple VM should be powered off before creating the Clone VM.

I also kept a pristine copy of the template VM so I would always have something to copy from when creating a new VM. For the sake of argument, let’s go with the following names and terminology so we can all stay on the same page. The template VM is going to be named Template. The cloned VM is going to be named Clone. I am going to assume that the template VM that you are using is already fully created, configured, and installed. I am also assuming that you either have console or SSH access to the host since you will need to have access to the commands on the computer itself.

The first step is to copy the template directory. My volume is named Array1, so the command looks like this (Note: I add the & to put the command in the background since it takes a while):

[root@vm1 ~]# cp -arp /vmfs/volumes/Array1/Template /vmfs/volumes/Array1/Clone &

Now its time to get started on the file editing. The first group of files we have to mess with are in the /etc/vmware/hostd/.

vmInventory.xml:
Assuming the only virtual machines you have are going to be Template and his buddy Clone, the following is what your vmInventory.xml should look like:


  
    32
    /vmfs/volumes/4725ae82-4e276b80-4c76-001c23c38d80/Template/Template.vmx
  
  
    48
    /vmfs/volumes/4725ae82-4e276b80-4c76-001c23c38d80/Clone/Clone.vmx

The 3 items that you have to note here are:

id: This is a 4 digit zero-padded number going up in increments of 1
objID: This is a number going up in increments of 16
vmxCfgPath: Here you need to ensure that you have the proper hard path (not sym-linked)

pools.xml:
Using the same assumption as before, the only 2 VMs are Template and Clone


  
    Resources
    ha-root-pool
    host/user
  
  
    2007-10-30T16:23:57.618151Z
    32
    ha-root-pool
    
      normal
      normal
    
  
  
    2007-10-30T16:23:57.618151Z
    48
    ha-root-pool
    
      normal
      normal

The 3 items that you have to note here are:

id: This is a 4 digit zero-padded number going up in increments of 1 (and it must match the id from vmInventory.xml
objID: This is a number going up in increments of 16 (and it must match the id from vmInventory.xml
The lastModified item here doesn’t matter as it will be changed when you make a change to VM anyway.

By now, the Template directory should be finished copying itself over to the directory that we will be using as our clone. First thing we have to do is rename all the files in the directory to mimic the name of our VM.

  # mv Template-flat.vmdk Clone-flat.vmdk
  # mv Template.nvram Clone.nvram
  # mv Template.vmdk Clone.vmdk
  # mv Template.vmx Clone.vmx
  # mv Template.vmxf Clone.vmxf

Now we just need to edit some files and we are ready to go. First let’s edit the Template.vmdk file. You need to change the line that reads something similar to (the difference will be in the size of your extents):

# Extent description
RW 20971520 VMFS "Template-flat.vmdk"

to look like:

# Extent description
RW 20971520 VMFS "Clone-flat.vmdk"

Save and exit this file. The next file is Template.vmx. The key here is to change every instance of the word Template to Clone. There should be 4 instances:

nvram
displayName
extendedConfigFile
scsi0:0.fileName

Don’t forget to change the MAC address(es). Their variable name(s) should be something like ethernet0.generateAddress. Delete the line that has the variable title sched.swap.derivedName. It will be regenerated and added to the config file. Lastly, add the following line to the end of the file if it doesn’t already exist elsewhere in the file:

uuid.action = "create"

The final item that needs to be done is the one that got me for such a long time. This is the step that will allow your changes to be seen in the client. (Drum roll …..)

Restart the VMWare management console daemon. So simple. Just run the following command:

  # /etc/init.d/mgmt-vmware restart

Note: This will log you out of the client console. But when you log back in, you will have access to the changes that you have made including the clones.

Good luck, and be careful as XML has a tendency to be easier to break than to fix.

Asterisk Caller ID Blocking Recipe

eric — Tue, 25 Sep 2007 13:36:28 +0000

Here’s another quick little Asterisk recipe that I threw together. It’s a handy because it only takes about 10 minutes to setup and is infinitely useful to the sales types. Just a note, this was done with Asterisk 1.4.8.

I wanted to do a little in AEL just to get a feel for it. It is a little AEL and a little regular extensions.conf type stuff.

The basic way that this CallerID Blocking recipe works is to use the Asterisk DB. An entry with the family of CIDBlock and the key of the dialing users extension will have a value of 1 or 0. The user initially sets their preference to either enabled (1) or disabled (0). When the number gets dialed, the preference gets checked and then the CALLERID(name)/CALLERID(number) values are set accordingly. In order for the user to enable CID Blocking, they need to dial *81. It will stay enabled until they dial *82.

How do we accomplish this? Easy. The sounds come with the asterisk sounds package.

Open up your extensions.conf and add the following lines (to whichever context works for you):

; Enable CallerID Blocking for the dialing extension
exten => *81,1,Set(DB(CIDBlock/${CHANNEL:4:4})=1)
exten => *81,2,Playback(privacy-your-callerid-is)
exten => *81,3,Playback(enabled)
exten => *81,4,Hangup()

; Disable CallerID Blocking for the dialing extension
exten => *82,1,Set(DB(CIDBlock/${CHANNEL:4:4})=0)
exten => *82,2,Playback(privacy-your-callerid-is)
exten => *82,3,Playback(disabled)
exten => *82,4,Hangup()

The last modification that needs to happen is that you have to change the exten that dials out to check the DB and react accordingly. Here is a snippet of mine (with the numbers changed to protect the innocent):

; Outbound calling for 111.222.3456 (my phone number)
exten =>_1NXXNXXXXXX,1,Set(CIDBlock=${DB(CIDBlock/${CHANNEL:4:4})})
exten =>_1NXXNXXXXXX,2,Set(${IF($[${CIDBlock} = 1]?CALLERID(name)=Unavailable:CALLERID(name)=MyCompany)})
exten =>_1NXXNXXXXXX,3,Set(${IF($[${CIDBlock} = 1]?CALLERID(number)=0-1-999:CALLERID(number)=1112223456)})
exten =>_1NXXNXXXXXX,4,DIAL(SIP/provider/${EXTEN},60,tr)
exten =>_1NXXNXXXXXX,5,Hangup

That’s really all it takes to set it up. Quick and handy.

* Note: *81 & *82 are arbitrary number combinations. Adjust to what works for you.

If you’re feeling really frisky, I added this AEL extension to check the status of your CallerID Blocking on *83. For fun, I have also included my *67 script for those who need an idea of how its done. As with almost anything in Asterisk, there are many ways to do it, this is just how I chose to accomplish this.

// Extra's for sending things outbound
context outbound-extra {
   *83 => {
            Playback(privacy-your-callerid-is);
            Set(CIDBlock=${DB(CIDBlock/${CHANNEL:4:4})});
            if(${CIDBlock} == 1) {
                Playback(enabled);
            }
            else {
                Playback(disabled);
            }
            Hangup();
   };

   *67 => {
      // Remove the *67 from the number we are dialing
      Set(dialed_number=${EXTEN:3:11});
      Set(CALLERID(name)=Unavailable):
      Set(CALLERID(number)=0-1-999):
      DIAL(SIP/provider/${dialed_number},60,tr);
      Hangup();
   };
};

Asterisk *69 with 1.4.x

eric — Mon, 02 Jul 2007 18:32:05 +0000

Many phone users just take for granted the service provided by the Telco called *69 (pronouced “Star six nine”). Since Asterisk is a telephony platform, it doesn’t just come with *69 built in. So if you want it, you have to implement it. To save you some time, I have implemented it with a few tweaks. This setup works, but YMMV (your mileage may vary).

The concept of the operation here is as follows: When a call comes in, we grab the caller id number and store it in the Asterisk DB. When a user makes an outgoing call to *69, we then get that last phone number that called in from the AstDB and dial it using our standard Dial() function. I will get deeper into each phase as I go through the process.

Just to make this all a little clearer, I will say that the context for making outgoing calls is outbound and the context for internal calls is internal.

1. The first thing to do is to modify the actions a call takes when it comes to the phone. Assuming the first line of your dialplan for the phone was:
exten => _1[0]XX,1,Dial(SIP/${EXTEN},21)
This would take the call and send it to which ever SIP peer matched (be it 1000, 1001, etc). To ensure that only non-internal calls get saved to our AstDB, I have added a statement to avoid calls coming from the internal context. This is our new step 1. If the call comes from the internal context, goto step 3.
exten => _1[0]XX,1,GotoIf($["${CONTEXT}" = "internal"]?3)
If not, continue on to our step 2. Here we are going to make use of the DB(family/key) function. (Note: For those who had trouble with this function (like me), the family is like a table and they key is like a column name). I set the family name to LastCIDNum and the key to be the receiving caller’s extension. The value was set to the callerid number. This was done as follows:
exten => _1[0]XX,2,Set(DB(LastCIDNum/${EXTEN})=${CALLERID(number)})
I then move the original Dial back to step 3. Our final internal product looks something like this:
exten => _1[0]XX,1,GotoIf($["${CONTEXT}" = "internal"]?3) exten => _1[0]XX,2,Set(DB(LastCIDNum/${EXTEN})=${CALLERID(number)}) exten => _1[0]XX,3,Dial(SIP/${EXTEN},21) exten => _1[0]XX,4,Voicemail(${EXTEN}@voicemail,u) exten => _1[0]XX,5,Hangup() exten => _1[0]XX,102,Voicemail(${EXTEN}@voicemail,b) exten => _1[0]XX,103,Hangup()

2. The next step is handle the outbound context for when a *69 call is placed. Assuming you don’t have an outbound dialing macro, we will handle this similarly to way an outbound SIP call would be placed. First we set the outbound callerid information:
exten => *69,1,Set(CALLERID(number)=12345678901) exten => *69,2,Set(CALLERID(name)=MyCompany)
Then we grab the last caller id information obtained. It would probably be a good idea to check if its there and if its not set to anonymous or something along those lines, but that is something that would be relatively easy to implement after the basics are up and running. To obtain the caller id information from the AstDB, I use the ${CHANNEL} variable to get the callers extension for the query. I use the substring variable syntax to pull the 4 digit extension out of the ${CHANNEL} variable. I then stick it in a temporary variable that I can lastcall.
exten => *69,3,Set(lastcall=${DB(LastCIDNum/${CHANNEL:4:4})})
Once we have that information, we can dial out almost as normal. The one issue is that for all US calls, it doesn’t receive the 1 in the callerid(num). So in the Dial function, I add a 1 for domestic calls.
exten => *69,4,DIAL(SIP/yourprovider/1${lastcall},60,tr)
(Note: The record in the CDR gets added as the outbound dialed number, not *69.)
Our final product for the outbound context should look something like this:
exten =>exten => *69,1,Set(CALLERID(number)=12345678901) exten => *69,2,Set(CALLERID(name)=MyCompany) exten => *69,3,Set(lastcall=${DB(LastCIDNum/${CHANNEL:4:4})}) exten => *69,4,DIAL(SIP/paetec/1${lastcall},60,tr) exten => *69,5,GotoIf(${DIALSTATUS} = CHANUNAVAIL,7) exten => *69,6,GotoIf(${DIALSTATUS} = CONGESTION,7) exten => *69,7,Hangup exten => *69,101,Congestion

In order to see if your DB() calls are working properly, you can run the command database show from the Asterisk console. It will find all the keys entered in the family “LastCIDNum”. If these are all (or mostly) external phone numbers, then you have likely done this setup correctly.

Configuring a Cisco 7961 for SIP and Asterisk

eric — Tue, 29 May 2007 19:39:28 +0000

Just prior to writing this, I think I was about ready to kill someone. Setting up this phone was probably one of the most challenging things I have done in a long time. So this will be my attempt to explain to other’s what I did and I will hopefully save some people some time.

Since we all need to be on the same page, let’s start out with the conventions:

Asterisk: Gentoo Linux, 192.168.1.5
Workhorse: Gentoo Linux (DHCP, TFTP, NTP), 192.168.1.20
Phone: Cisco 7961
Anything starting with a $ means you put your value in it. I will name the variable something descriptive for you
Remember that all filenames with Cisco are case sensitive
If there are some files you need examples of or access to and aren’t listed, please don’t hesitate to contact me.

I am not going to go into a lot of detail with things, just give some overview and some examples and it will hopefully be enough to get you in the right direction. Check throughout the document for some references and read up on those if need be.

DISCLAIMER: I am not an expert. If you break your phone while doing anything I mention here, I am not responsible. This is just what I did to get everything to work.

1. The first order of business was to add the phone’s MAC address to DHCP so I could be sure what was accessing the tftp server. I also needed to know the MAC address to create the proper files in the tftp directory. Ensure that you set the tftp server, ntp server, and SIP server in DHCP.

group voip {
        option domain-name-servers 192.168.1.20, 1.2.3.4;
        option domain-name "inside.mycompany.com";
        option smtp-server 192.168.1.20;
        option ntp-servers 192.168.1.20;
        option time-servers 192.168.1.20;
        option routers 192.168.1.1;
        option sip-server 192.168.1.5;
        default-lease-time 86400; # 1 day
        max-lease-time 86400;
        server-name "192.168.1.20";
        option tftp-server-name "192.168.1.20";

        host myphone {
            hardware ethernet 00:19:E8:F4:B4:D0;
            fixed-address 192.168.1.200;
        }
}

2. When you first plug in the phone, it’s loaded with the Skinny protocol software only (SCCP), nothing for SIP. This is because the phone was designed to work best (and really only) with the Cisco Call Manager. The first thing I had to do was to obtain the files that go in the tftproot on 192.168.1.20. In the upgrade package were the files:

apps41.1-1-3-15.sbn
cnu41.3-1-3-15.sbn
copstart.sh
cvm41sip.8-0-3-16.sbn
dsp41.1-1-3-15.sbn
jar41sip.8-0-3-16.sbn
load115.txt
load30018.txt
load308.txt
load309.txt
SIP41.8-0-4SR1S.loads
term41.default.loads
term61.default.loads

3. Once you place these files in the tftp root directory, you are ready for the upgrade. (Note: You need a Cisco smartnet file (or be good with Google) to find these files). Upgrading requires a factory reboot of the phone so it will look for the term61.default.loads file. To perform a factory reset of the phone, hold down the ‘#‘ as the phone powers up. Then dial ‘123456789*0#‘ and then let it work. The next time it reboots, it should then grab the necessary files from the tftp server and upgrade itself. You can watch the tftp logs and the phones LCD to ensure that everything that is supposed to be happening is happening.

4. At this point, the phone should be able to completely boot up and will likely just show you the word Unprovisioned at the bottom of the screen. The next step is to create the files that each phone needs to survive. The first file we are going to create is the SEP$MAC.cnf.xml. In the case of the phone that I am going to use for this demo, the filename is: SEP0019E8F490AD.cnf.xml. I know that the phone is also requesting the file CTLSEP0019E8F490AD.tlv, but you can safely ignore that. The minimalist version of the SEP$MAC.cnf.xml file:


   SIP
   cisco
   cisco
   
      
         M/D/Ya
         Eastern Standard/Daylight Time
         
              
                  192.168.1.20
                  Unicast
              
         
      
      
         
            
               
                  
                     2000
                     5060
                     5061
                  
                  192.168.1.5
               
            
         
      
   
   
      
         
         
         
         
         
         
         true
      
      
         true
         x--serviceuri-cfwdall
         x-cisco-serviceuri-pickup
         x-cisco-serviceuri-opickup
         x-cisco-serviceuri-gpickup
         x-cisco-serviceuri-meetme
         x-cisco-serviceuri-abbrdial
         false
         2
         true
         true
         2
         2
         1
         true
      
      
         6
         10
         180
         3600
         5
         120
         120
         5
         500
         4000
         70
         true
         None
      
      1
      false
      true
      false
      false
      g711ulaw
      101
      3
      avt
      false
      false
      3
      0
      
      LinkExperts
      1
      true
      10
      false
      16384
      32766
      
         
            9
            100
            192.168.0.205
            5060
            100
            Eric Lubow
            
               2
            
            3
            100
            
            false
            1
            *97
            4
            5
            100
            
               true
               true
               false
               true
            
         
      
      5060
      184
      0
      dialplan.xml
   
   
      
      true
      1
   
   SIP41.8-0-4SR1S
   
      false
      false
      1
      1
      0
      0
      0
      0
      1
      1
      1
      
   
   1143565489-a3cbf294-7526-4c29-8791-c4fce4ce4c37
   US
   
      US
      5.0(2)
   
   1
   
   
   
   
   
   proxy:3128
   
   96
   0
   96
   4
   0
   
      
         3804
      
   
   
   false

5. You will also need to create a dialplan so the phone doesn’t try to dial immediately. Below is a minimalist dialplan.xml (which is the filename we used in the above schema).