VoIP PBX

The iPAQ 514 is a next generation mobile product from HP. Called "Voice Messenger", this smartphone is distinguished both for its size (107x49x16.3m-HxWxD) and weight (102gr.) for the technical features: tri-band GSM + EDGE, Wi-Fi 802.11b / g, Display 2 " from 176 × 200 65k color, 1.3MP camera, Windows Mobile 6, support the SIP protocol.

Hp Ipaq 514 The configuration is done through a fairly simple utility - available on CD in the package of the phone - named "HP iPAQ Setup Assistant" and the phone connected via the USB cable to your PC.

Before you make the appropriate configurations, I updated the firmware with the newest release, 05/02/00, available in the HP support site .

At this point, after configuring the Wi-Fi, we proceed to the SIP configuration:

Image 1	Image 2	Image 3

In the three images are represented here at the top the three steps for the activation of the configuration. In image 2, in particular, depicts the sample configuration file that we will use to connect the iPAQ 514 to our asterisk server. Once all the parameters, and ended on the HP iPAQ Setup Assistant wizard, the configuration will be transferred to your phone.

We must pay attention to one detail: when you attach your phone to USB cable for PC connection, the Wi-Fi smartphone is automatically disabled. After finishing the configuration, you must unplug your phone from your USB cable!

The configuration in the file sip.conf of asterisk, is the following:

[300]
type=friend
secret=300
qualify=yes
port=5060
nat=yes
mailbox=300@device
host=dynamic
dtmfmode=rfc2833
disallow=
dial=SIP/300
context=from-internal
canreinvite=no
callerid=device <300>

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Chan_mobile is a channel driver for Asterisk, written by David Bowerman, allowing you to use Bluetooth as FXO devices Phones and Bluetooth Headsets as FXS devices.

The main features are:

• Most phones can be connected (subject to certain limitations)
• Support for several Bluetooth adapters
• Asterisk automatically connects to each cell when it enters the field of Bluetooth.
• Incoming calls on mobile phones are handled by Asterisk as normal calls on Zap channels.
• Outgoing calls on mobile phones using the Command Dial (CELL / device / nnnnnn) in the dialplan.
• Ability to use a Bluetooth Headset as extension using the Command Dial (CELL / device) in the dialplan.
• The application CellStatus can be used in the dialplan to check whether a mobile phone is connected.
• MobileSMS application to send SMS from a phone connected.
• Support devicestate.

Fully Supported Phones:

• Nokia 6021
• Nokia 6230i
• Nokia E51

Partially supported phones:

• LG TU500
• LG CU500
• RIM Blackberry 7250
• VK 2020
• Sony Ericsson T series T68, T68i, T300, T310, T610, T630
• Sony Ericsson K series k700i
• Sony Ericsson v600i V series
• Motorola L6
• Motorola V3
• Motorola V195
• Nokia 6310i
• Nokia 7600
• Palm Treo 650
• NOKIA 6111
• NOKIA 6830
• NOKIA 6233, 6234
• Samsung SGH-E720 (voice only)

Links:

• http://www.chan-mobile.org/
• http://svn.digium.com/view/asterisk-addons/trunk/chan_mobile.c
• http://www.geek-pages.com/articles/asterisk/howto_build_and_configure_chan_mobile_on_trixbox.html
• http://www.voip-info.org/wiki/view/chan_mobile

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The Asterisk development team has released Asterisk 1.6.0 beta 1. The community effort is required to help in testing Asterisk 1.6, so that the final release will take place as soon as possible.

Asterisk 1.6 will be a version change and therefore a "major release" compared to version 1.4, which was released a year ago. The new version contains new features and changes to the internal architecture of the system that will improve the performance of the phone system.

A list of all the new features is available in the file:

CHANGES

Asterisk 1.6 will also bring a new mode of release management. These new policies on the release are the result of experience that the team has learned asterisk during the update of Asterisk 1.2 and 1.4.

In any event, will remain the support versions 1.2 and 1.4 that will not change in the immediate future.

Links:

• Download Asterisk 1.6.0 beta 1

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The word Zaptel is the diminutive of "Zapata Telephony" and represents the driver architecture originally written by Jim Dixon to give telephone support to its hardware platform of BSD. Digium has subsequently produced its phone cards starting from the design of Jim Dixon making a port of the driver from BSD to Linux. Currently, the zaptel driver, there is support for many hardware phone cards.

Recently, the development team has rilasciatoun Asterisk Zaptel driver update to version 1.2.23 and 1.4.8.

The release contains a series of new bugs and fiexs caratteistiche, such as:

• Improvements to the utility fxotune
• Full support for Digium cards: TE120P, TE121P, TE122P
• Updates to the generation module now allows DTMF tone generation at runtime, as well as support for a DTMF twist based on zones.

The release is available at:

• http://downloads.digium.com

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The Asterisk development team has released a new version 1.4.17: This release includes the fix for a security problem in SIP and another set of bugfixes.

The flaw in the security was issued in security advisor AST-2008-001. The vulnerability is the ability to crash the SIP channel driver with a transfer made ad hoc and is only present on systems Asterisk 1.4 and not even on versions 1.2.

The security advisor is available at: http://downloads.digium.com/pub/security/AST-2008-001.pdf .

The 1.4.17 release can be downloaded from the site of Digium.

References

• http://downloads.digium.com/pub/telephony/asterisk/ .
• http://downloads.digium.com/pub/security/AST-2008-001.pdf .

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The new version of Trixbox CE has been released and is based on CentOS 5.1 and Asterisk 1.4: this is the greatest feature of this release than previous.

One of the main problems of previous versions was the lack of hardware support for new systems, especially for Dell platforms. All this has now been resolved with version 2.4 and is now based on the latest CentOS 5.1 kernel, which provides support to many motherboards, network cards, and hardware such as RAID controllers.

Another important new feature is the presence of "high resolution timers" in-kernel: in this way is to drop the need for a timing mechanism on the PCI bus (eg that of the USB bus) making it easier to use in Trixbox virtualized systems such as Xen or VMWare.

The use of Asterisk 1.4 Trixbox makes available in the following characteristics:

• Generic JitterBuffer - improves the quality of a call in case of network congestion.
• AEL Version 2 - simplifies programming Asterisk Extensions Language and configuring the dialplan.
• T.38 FoIP protocol-management pass-through.
• Jabber/Jingle/GoogleTalk- protocol-level compatibility with networks Jabber, Jingle, and Google Talk.
• Increased language capabilities-improved support for English, Spanish and French (introduced new sounds and improved support to the structure of words).
• "Unified Messaging" - voicemail, email, and fax services are centralized in a single mailbox where users can send, receive and manage all their messages using any communication devices provided by the system. * Enhanced support of DTMF standard rfc2833
• Fax Support - Fax Support also improved, especially for users of analogue or digital.

References:

• Trixbox Homepage
• Trixbox 2.4 release

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Asterisk developers have taken measures to provide an update to both branch 1.4 and 1.2: in fact, the release contains the solution to a security flaw in the system. The 1.4.16 release also contains a number of bug fixes introduced in recent weeks.

The details of the security breach were published in a security advisory:

AST-2007-027.pdf
The issue affects users who use the dynamic realtime configuration method for IAX2 and SIP makes use of host-based authentication.

The full list of changes is available in the ChangeLog.

Related Links:

• Changelog 1.4.16
• Digium Downloads

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In the two units so far I've set up, the starting point was to update the firmware. The easiest way to do this is to download the latest firmware from the appropriate page of the site GrandStream : the latest available at the time of writing this article is 1.0.1.2.

We will make sure that you have a webserver which make our unity GXW-4104 can download the firmware. We'll have to unpack the zip with the firmware in a subdirectory of your webserver, for example in a folder called "firmware /". Assuming that the server responds to the IP 192.168.0.1, and opening a browser, we must ensure that - by going to http://192.168.0.1/firmware is displayed a list of the contents of the folder extracted previously, namely:

• boot64.bin
• boot64a.bin
• gxw4100.bin
• load64.bin

Without these basic steps, we will configure our unit so that it can take the firmware from our web server. By accessing the administration panel via our trusty browser notification, we will:

1. Activate the upgrade through HTTP protocol;
2. Set the path of the firmware ("Firmware Server Path" should be filled with the value - following the example given above - "http://192.168.0.1/firmware")
3. Enable auditing for the upgrade (Always check for New Firmware) and the date by which the check must be made ("check for upgrade every"): the lowest possible value is 60 minutes.

Without this, after 60 minutes set the firmware will be downloaded from the web and the next time unit we find the latest release of firmware you have time to grab a coffee and read the latest news on voip-PBX

After updating the firmware, do not forget to check by "Status" that the version of firmware is actually equivalent to that discharged.

Without the upgrade, we can do the configuration itself. We list the first steps:

1. Set the FXO ports and parameters of the PSTN tones
2. Set the SIP Server
3. Configure Asterisk

The configuration shown in this mini-tutorial aims to help you:

• receive calls on any port FXO and dispatch them - all - into a single sip account
• make calls using the PSTN directly from your SIP phones

1. Setting FXO ports
In "FXO Lines":

Under "FXO Termination"

• Enable Current Disconnect: Yes
• Enable Tone Disconnect: Yes
• Enable Polarity Reversal: No
• AC Termination Impedance: 270 Ohm + (750 Ohm | | 150 nF) and 275 Ohm + (780 Ohm | | 150 nF)
• Unconditional Call Forward to VOIP:
  • Username: ch1-4: 111; (It means that for channels 1 to 4 calls are redirected all'extension SIP/111 the sip server specified for each channel)
  • SIP Server: ch1-4: p1 (It means that for channels 1 to 4 should be used to configure the sip server specified in p1 = Profile number 1)
  • SIP Destination Port: ch1-4: 5060; (It means that for channels 1 to 4, the sip server is on port 5060)

In "Channel Dialing":

• Wait for Dial-Tone (Y / N): ch1-4: N;
• Stage Method (1/2): ch1-4: 1;

In the "Channels":

Under "Call Progress Tones":

• Dial Tone: ch1-4: f1 = 425 @ -14, f2 = 425 @ -14, c = 20/20-60/100;
• Ringback Tone: ch1-4: f1 = 425 @ -14, f2 = 425 @ -14, c = 100/400;
• Busy Tone: ch1-4: f1 = 425 @ -14, f2 = 425 @ -14, c = 20/20-20/20;
• Reorder Tone: ch1-4: f1 = 425 @ -14, f2 = 425 @ -14, c = 20/20-20/20;
• Confirmation Tone: ch1-4: f1 = 425 @ -14, f2 = 425 @ -14, c = 20/20-20/20

Under "Specific Channel Setting":

• DTMF Methods (1-7): ch1-4: 2;

2. Set the SIP Server

We will use a single profile, under "Profile 1", set the IP of your Asterisk server is in "SIP Server" in "Outbound Proxy".

3. Configuring Asterisk

Edit the file sip.conf:

[gxw410x]
type=peer
context=from-grandstream
host=ip_di_asterisk
insecure=port
dtmfmode=rfc2833

[111]
type=friend
secret=111
qualify=yes
nat=no
host=dynamic
canreinvite=no
context=gxw410x

Edit the extensions.conf file and add a rule for outgoing calls to the GXW 4104:

[grandstream]
exten => _0X.,1,Dial(SIP/${EXTEN:1}@gxw410x,30,r)
At this point you just have to modify your dialplan to handle incoming calls according to your needs.

This guide is not exhaustive and is an indication of the main steps you should take a basic configuration of this product. Possible configurations are much more advanced, but I leave this task to you: When you're familiar with the structure of the configuration panel, everything should become easier!

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At the mention of Call Center and Contact Center immediately came forward, in our imagination, the idea of ​​a worker with a bonnet and the phone at his side. In fact, however, call centers are far more complex reality: the definition of wanting to quote Wikipedia : "To call centers (or call center) means all equipment, computer systems and human resources to manage, in an optimized way, the telephone calls to and from a company. The activity of a call center can be performed by skilled operators and / or receptionists interactive IVR. The operators and receptionists can provide them, enable services, provide technical assistance, offering reservation services, enabling purchases and organize promotional campaigns (telemarketing). "

And in fact in a call center there is a data processing structure that is composed at least by:

• A phone server that handles channels and telephone calls to call centers, receptionists (IVR) , operators. Often a PBX or directly to the telephone lines (in this case the server can also operate as PBX phone, although the two concepts are distinct and PBX telephony server).
• A corporate LAN more or less articulate.
• A series of workstations with phone (hardware or software) and headset.
• We can find locations on a PC (Personal Computer) through which the operator performs dataentry or retrieves information from corporate databases to provide the client with whom you talk.
• In some specific cases we can find a system of CTI (Computer Telephony Integration), ie a system that allows the server to communicate with your phone and PC operator trasferirgli call information input / output.

Usually the call center communicates through telephone channels. And this is the fundamental distinction with the Contact Center. Taking another definition WikiPedia : "The contact center is a call center evolved, combining the functions of telecommunications with information systems, adding the use of other instruments over the telephone / communication channels, such as: the physical counter, mail , fax, mail, web, messaging on mobile phones. The call center, as well as the most advanced contact center, must be understood as a new way of managing contacts and relationships with customers and citizens, according to a broader strategic vision and planning time to tell a customer-oriented culture of the Administration "

So the Contact Center is a broader concept, for which you require further technological resources such as:

• SMS Server: a server that is capable of sending SMS over the GSM network. Can be achieved when using special software and GSM modems, or purchased in the form of outsourced service by specific providers.
• FAX Server: a server dedicated to sending fax through normal telephone lines. The FAX Server become indispensable tools for work when the call center sells a service that sends faxes to customers: the automation of transmission makes it easier and cheaper labor, as it prevents the operator to worry about sending the fax (or resend in case of no transmission). The fact FAX Server tools are fully managed by the IT department and therefore controllable by the network.

Often the call center make use of special software for contact management that, in technical jargon, are called CRM (which stands for Customer Relationship Management, which is software for managing customer contacts). Through this software you can store disparate information about customers of the call center. The CRM concept is fairly large, but tends to its main functions are:

1. The acquisition of new customers (or "potential customers")
2. Increased relationships with key customers (or "customers culturable")
3. The longest possible retention of customers who have more dealings with the company (called "clients first floor")

In the call center making outbound activities (ie outgoing calls) you can find the software dedicated to this type of contracts denominated CATI (which stands for Computer Aided Telephony Interview Software to manage the computer of telephone interviews).

With this overview, certainly not exhaustive, we tried to give a first overview of the technological complexity of structures needed to run a call center: it is often heterogeneous systems that require integration and development of dedicated software, and in some cases even extremely expensive.

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VAD, an acronym for Voice Activity Detection, is an algorithm used in the preparation voice to detect the presence or absence of human voice in the audio championship. Usually, the VAD is used in audio coding and speech recognition systems.

Citing a nice stretch from Cisco website : "in conversations at a normal, if one speaks the other listens. In today's communication networks are used bidirectional channels at 64kbps regardless of whether someone is speaking. This means that is wasted at least 50 percent of the total available bandwidth. The amount of wasted bandwidth can be even higher if we analyze statistically how many there are pauses in speech in a conversation media. "

Just to save banda VAD, already in use on GSM networks, was introduced by some national carrier VoIP.

Of course, the VAD has created a problem where there is an installation of Asterisk, especially if you have an auto responder (IVR) that makes use of music on hold (Music on Hold): Asterisk is not normally able to generate packets RTP output if the incoming packet does not arrive due to the removal of silence. End result, the hold music is heard at times or rather only when the other side of the handset is channeled voice or some background noise ... The problem came to my attention during a testing phase of an IVR, with the invaluable assistance of his friend Alfredo Gentile that I take this opportunity to say goodbye.

To overcome this problem, in Asterisk 1.4 has been introduced - already for some time and after some testing of various applications available - a patch that fixes the problem. In practice, forces the generation of outgoing packets asynchronously when there is a source of external timing: If on the other end there is a source of precisely timing Asterisk will generate outgoing packets asynchronously.

It will therefore be sufficient to ensure:

1) Have a release of Asterisk 1.4

2) Load the module ztdummy if you do not have a card that makes use of zaptel and make sure that the startup is Automatic boot.3) Add the following lines to / etc / asterisk / asterisk.conf:

[options]
internal_timing = yes
silence_supression=no

For those using XEN, is also needed to recompile zaptel, after changing the code of the source file ztdummy.c comment out the lines 57 to 70:

/*#if defined(__i386__) || defined(__x86_64__)
#if LINUX_VERSION_CODE >= VERSION_CODE(2,6,13)
The symbol hrtimer_forward is only exported as of 2.6.22:
#if defined(CONFIG_HIGH_RES_TIMERS) && LINUX_VERSION_CODE >= VERSION_CODE(2,6,22
)
#define USE_HIGHRESTIMER
#else
#define USE_RTC
#endif
#else
#if 0
#define USE_RTC
#endif
#endif
#endif
*/

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