Save Lono

This tutorial describes how to configure the idle display screens on Polycom phones to subscribe to a XHTML file that reads their Agent status from Asterisk.  When the phone is not in active use it displays the web page.

My tutorial has three components.  A specific Polycom option(found in sip.cfg) added to the configuration file of each phone, a BASH script I wrote, and a little Asterisk configuration.  This tutorial assumes you have a working Asterisk 1.6+ PBX with a basic queue setup.  It also assumes you are using automatic web provisioning of your Polycom phones.

Here is a link to my previous article on  configuring Polycom SoundPoint IP phones for Asterisk:

How to configure a Polycom SoundPoint IP phone for Asterisk on Fedora 10

To get the idle display screen to work correctly we will need to modify our standard configuration file a bit.  Navigate to the FTP/HTPP directory you provision your phones from.  Edit an Agent phone similar to the following,
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I’m knee deep into writing Voice Provision .04(VP4) for VoiceIP Solutions.  Why write another Asterisk GUI?  Because the Trixbox and Elastix software is way to convoluted and makes customization difficult.  They are not well documented and don’t attribute credit back to Asterisk or Digium.

Standard Asterisk scripting and basic macro’s are easier to setup, maintain, develop and troubleshoot.  What I require is a basic PHP SIP/IAX2 user manager that stays out of the dial plan. It must be specifically designed to manage Polycom phones.  This is why I’ve been locked in my room the last 4 weeks and haven’t done any recent articles.  My apologies.

The goals for VP4 are simple, Dynamic realtime SIP user management and standard Asterisk dial plan work via the extensions.conf.  Since VoiceIP Solutions is a certified Polycom reseller, VP4 will include several functions for generating Polycom configuration files and allowing remote reboots of the phones.  In this post I will provide a example of PHP function that reboots a Polycom SoundPoint IP phone by sending a SIP Notify message.

This post assumes you have a basic grasp on LAMP(Linux Apache MySQL PHP).  Also you need a working registered Polycom phone provisioned via a boot server of some kind(HTTP,FTP, TFTP,etc…).  I have a tutorial on that here.

Goals of this Post:
- Install PHP-SIP class from Google Code
- provide example function in PHP

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A customer of VoiceIP Solutions sent me this useful tutorial for re-sampling Polycom Ringtones.   If you have an MP3 you like, it can be re-sampled for use with a Polycom IP SIP phone!  The procedure is pretty straight forward, Install Audacity with yum or your favorite package manager, re-sample the track, then edit the proper Polycom Soundpoint boot files.

Goals of this Post:

- Install Audacity Digital Audio Editor
- Convert a MP3 to Polycom compatible track
- Enable custom special Ringtone on Polycom Soundpoint IP SIP phone

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The Digium Asterisk Appliance has built-in features for handling Polycom SIP handsets.  It’s very easy to do and will save you the trouble of individually configuring the settings of each phone.  I have a Digium AA50 configured with a standard dial plan and a Polycom Soundpoint IP 501.

Goals of this Post:

- Configure AA50 Polycom Provisioning
- Configure Polycom SoundPoint IP 501 phone

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This file is named by the MAC address of your Polycom SoundPoint IP SIP phone followed by, ‘-phone.cfg’.  In your FTP folder you would have a file for each phone – I have a single phone.  It’s MAC is ’0004F202734B’; so my phones configuration file would be named, ’0004f202734b-phone.cfg’.  I believe there are other conventions for naming this file as well.

The example below was used to connect my phone with a VoiceIP Solutions Asterisk PBX.  This example shows just a fraction of the many possible features in this line.  For my purposes, I defined the Asterisk server IP address, and it’s SIP credentials.  I also added the NTP server.  The ‘mwi’ tag refers to ‘message waiting information’, here I set the mailbox(s) I’m subscribing to and the extension to check voicemail.   My Asterisk voicemail menu is extension ’299′.

0004f202734b-phone.cfg:

<?xml version=”1.0″ encoding=”UTF-8″ standalone=”yes”?>
<!– Example Per-phone Configuration File –>
<!– $RCSfile: phone1.cfg,v $  $Revision: 1.104.2.2 $ –>
<phone1>
<reg
reg.1.displayName=”5555″
reg.1.address=”5555″
reg.1.auth.userId=”5555″
reg.1.auth.password=”2005″
reg.1.server.1.address=”192.168.1.254″
tcpIpApp.sntp.address=”pool.ntp.org”
tcpIpApp.sntp.gmtOffset=”-33600″
>
<mwi
msg.mwi.1.subscribe=”5555″
msg.mwi.1.callBackMode=”contact”
msg.mwi.1.callBack=”299″
>
</phone1>

I was working on a project for VoiceIP Solutions yesterday and I required a FTP server to provision some Polycom Soundpoint IP phones. So I did my usual ‘yum install proftpd’ , followed by ‘service proftpd start’, but the service failed to start. So after some poking around – checking user assignment for that process and permissions I found nothing. So I decided to invoke the server from the command line with the ‘-t’ to check for errors:

[root@mattcom1 log]# proftpd -t
Checking syntax of configuration file
– warning: unable to determine IP address of ‘mattcom1′
– error: no valid servers configured
– Fatal: error processing configuration file ‘/etc/proftpd.conf’

The error told me right away that the issue was my ‘/etc/hosts’ file. My server name ‘mattcom1′ was not found in that file. Even though both the Network manager and Network configuration tools showed a hostname, there was actually nothing added to the ‘/etc/hosts’ file. watch out for this one!

I told a customer for the company I for that I would figure out how to logout agents by CID(Caller ID). So I figured, why not kill two birds with one stone. Today we will create a single Queue, Agent, and dial plan to accomplish this goal. I’m using Asterisk 1.4, Fedora 10 and a Polycom IP SIP phone for my demonstration purposes.

When I started this project four hours ago, I thought I would google my way to another successful blog post(and happy customer), but no…  logging out agents in Asterisk is very unintuitive.  The agentcallbacklogin utility has the exact same prompts for logging in as out.  AgentCallbackLogin (when initiated, from the dial plan)  asks for three things, agent, agent password, and call back number.  To eliminate all these prompts I’m using the ‘$CallerID(num)’ variable to automatically answer the agent and call back number.  So the user 8888 dials ’1000′ and and AgentCallbackLogin assumes he is AGENT/8888 with a password of ’8888′.

from /etc/asterisk/extensions.conf

exten => 1000,1,AgentcallbackLogin(${CALLERID(num)}||${CALLERID(num)}@savelono-queue-out)
exten => 1000,n,hangup

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What makes Asterisk so great is the growing ecosystem of 3rd party software.  For call centers based on Asterisk PBX, the best on the market is Queuemetrics.  This solution allows for over 150 different statistics to be collected.  Here is just a few:

Number of calls
Total call length
Average call length
Average call waiting
Number of unanswered calls
Average time before disconnection
Area code
Number of calls
Total calling time
Average time per call (for taken calls)
Average wait per call
Average position at disconnection (for lost calls)
Number of available agents
Total agent time
Average agent time
Minimum/ maximum agent session duration
Agent availability

If by now you are not convinced take a look at the complete list; you can check out Loway’s site and Queuemetrics here.  I have personally helped setup several commercial call centers(while working for VoiceIP Solutions) with this software and I am impressed at the value and support for our customers purchase.  So today we are going to set up a basic Queuemetrics installation; we will not be covering the Asterisk portion.  Nothing fancy here, but I can get you started.

The goal of this post:

- install Queuemetrics

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In my opinion the best IP business phones on the market are made by Polycom. Anyone that knows anything about the VoIP Industry knows that!  High quality Polycom desk phones combined with Asterisk are a great combination of quality/price. So to that end we’re doing this lab.

Polycom employs several methods of provisioning the SIP phones.  For general configuration Sound Point IP have an excellent built web GUI,  but for multiple phones Polycom has an XML based system as well.  Every Sound Point IP can be provisioned based on MAC address.  Polycom’s provisioning method makes use of TFTP, FTP, or HTTP to deliver firmware updates and individual phone settings.

The goals of this post:

- Configure FTP server for Polycom firmware and configuration

- Configure Asterisk SIP extension

- deploy firmware and XML configuration files to Polycom SoundPoint IP 501 SIP phone

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VoiceIP Solutions is a Asterisk ‘consulting & deployment’ company in Seattle Washington.  They deploy Asterisk solutions for businesses of all sizes.  From small offices to universities and call centers.  They have sites deployed all over the United States, but mostly on the West Coast.  I’ve been following them for some time; I guess they started deploying Asterisk before 1.2 was released.  I talked to one of their sales rep’s(I think his name was Liam) about the business and wondered if they had done any development work?  He told me that they had done some PHP work for managing larger installs and proceeded to direct me to one of there engineer/developers.

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