VoIP PBX

Asterisk developers have managed to release an update for both branch 1.4 and 1.2: the issue has in fact 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 flaw has been published in a security advisory:

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

The entire list of changes is available in the ChangeLog.

Related links:

• ChangeLog 1.4.16
• Downloads Digium

Tags: Call Center Systems | VoIP PBX | Asterisk Consultant Naples | PBX | Voip | Asterisk CTI | PBX | IP Phones | Networking | Linux

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Development of IVR systems, callcenter, PBX Voip.

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

We make sure to have a webserver which make our unity GXW-4104 can download the firmware. We unpack the zip with the firmware in a subdirectory of your webserver, eg in a folder called "firmware /". Assuming that the server responds to IP 192.168.0.1, and opening a browser, we must ensure that - logging at http://192.168.0.1/firmware is displayed listing the contents of the extracted folder earlier, namely:

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

Without these basic steps, go to configure the unit so we can remove the firmware from our web server. By accessing the administration panel via our trusty browser requirements:

1. Enable upgrade via HTTP;
2. Set the path of the firmware ("Firmware Server Path" should be filled with the value - following the example done before - "http://192.168.0.1/firmware)
3. Enable auditing to upgrade (Always check for New Firmware) and the period within which the check must be done ("Every check for upgrades"): the lowest value possible is 60 minutes.

Without this, after 60 minutes set the firmware will be downloaded from the web and restart the unit we find the latest firmware release: you have the time to have coffee and read the latest news on PBX-VoIP

After the firmware update, do not forget to check the "Status" that the firmware version 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 PSTN tones
2. Set the SIP Server
3. Configuring Asterisk

The configuration described in this mini-tutorial aims to enable you to:

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

1. Setting FXO ports
In "FXO Lines":

Under "Termination FXO"

• Enable Current Disconnect: Yes
• Enable Disconnect Tone: Yes
• Enable Polarity Reversal: No
• AC Termination Impedance: 270 Ohm + (750 ohms | | 150 nF) and 275 Ohm + (780 ohms | | 150 nF)
• Unconditional Call Forward to VOIP:
  • Userid: ch1-4: 111; (It means that for channels 1 to 4 calls will be forwarded all'extension SIP/111 the SIP server specified for each channel)
  • SIP Server: ch1-4: p1 (It means that for channels 1 to 4 will 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 (half): ch1-4: 1;

In "Channels":

Under the 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 one profile, under "Profile 1", set the IP of your Asterisk server is "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 adjust 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. Configurations are much more advanced, but I leave you this task: When you got to know the structure of the configuration panel, all should become easier!

Tags: Call Center Systems | VoIP PBX | Asterisk Consultant Naples | PBX | Voip | Asterisk CTI | PBX | IP Phones | Networking | Linux

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Development of IVR systems, callcenter, PBX Voip.

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To hear the Call Center and Contact Center immediately comes forward, in our imagination, the idea of a worker with a cap and the phone at his side. In fact, however, call centers are far more complex reality: Those wishing to cite the definition of Wikipedia : "To call centers (or call center) means all equipment, information systems and human resources to manage, optimally, phone calls to and from a company. The activity of a call center can be carried out by skilled and / or responder interactive IVR. Operators and answering machines can offer information, activated services, technical assistance, offering reservation services, allow purchases and promotional campaigns (telemarketing).

And in fact in a call center there is a computer structure that comprises at least:

• A phone server that handles the phone network and calls to call centers, the responder (IVR) , operators. Often a PBX or directly to telephone lines (in this case the server can act as a PBX telephone, despite the two concepts are distinct and PBX phone server).
• A corporate LAN more or less articulated.
• A series of workstations equipped with telephone (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 cases we find a system of CTI (Computer Telephony Integration), ie a system that allows the server to communicate with the telephone operator's computer and transfer information to the call in / out.

Traditionally, call centers communicate through telephone channels. And this is the fundamental distinction with the Contact Center. Taking another definition of WikiPedia : "The Contact Center is an advanced call center, combining the functions of telecommunication, information systems, adding the use of means other phone / communication channels such as physical door, 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 method for 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 "

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

• SMS Server: a server that is able to send SMS via 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 faxes via the normal telephone lines. The FAX Server become indispensable tools to 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 faxes (or resubmit in case of failure). The FAX Server tools are in fact completely managed by the IT department and therefore controllable by the network.

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

1. Acquiring new customers (or potential customers)
2. The increase of relations with major customers (or customers arable)
3. The longest possible retention of customers who have more dealings with the company (as defined "customers first floor)

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

With this view, certainly not exhaustive, we tried to present a vision of the technological complexity of structures needed to run a Call Center: this is often heterogeneous systems that require integration and development of dedicated software, and in some cases extremely expensive.

Tags: Call Center Systems | VoIP PBX | Asterisk Consultant Naples | PBX | Voip | Asterisk CTI | PBX | IP Phones | Networking | Linux

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Development of IVR systems, callcenter, PBX Voip.

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

Citing a nice stretch from Cisco site : "In normal voice conversations, if one talks the other listens. In today's communications networks are used bi-directional channels at 64kbps whether someone is speaking. This means that it 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 pauses in speech are in a conversation media. "

Just to save bandwidth 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 automated (IVR) that uses a music on hold (Music On Hold): Asterisk is not normally able to generate RTP packets output if the incoming packets do not arrive due to the removal of silence. End result, the music is heard the waiting line or better on the other side only when the handset is channeled voice or any noise ... The problem came to my attention during a testing phase of IVR, with the invaluable collaboration of Alfredo Gentile that I take this opportunity to say goodbye.

To overcome this problem, Asterisk 1.4 was introduced - already for some time and after some testing of various applications available - a patch that fixes the problem. In practice it is forced to generate outgoing packets asynchronously when there is a source of external timing: If on the other end is not a source of precisely timing Asterisk generate outgoing packets asynchronously.

It will therefore be sufficient to ensure:

1) Have a release of Asterisk 1.4

2) Load module ztdummy, if you do not have a card that uses zaptel and ensure that the boot is the automatic boot.3) Add the following lines to / etc / asterisk / asterisk.conf:

[options]
internal_timing = yes
silence_supression=no

For those who make use of Xen, it is also necessary to rebuild zaptel, after modifying the source code files ztdummy.c commenting 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
*/

Tags: Call Center Systems | VoIP PBX | Asterisk Consultant Naples | PBX | Voip | Asterisk CTI | PBX | IP Phones | Networking | Linux

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Development of IVR systems, callcenter, PBX Voip.

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QueueMetrics is a complete management software for call centers based on Asterisk PBX. And 'the news of the December 4 issue - three months after 1.4.2 - new version 1.4.3.

This new version not only contains a number of bug fixes but also a host of new features and improvements. Reports have been introduced and has been completely renovated the configuration editor.

Among the new features are listed:

• New editor for users, classes, queues, agents, locations, calling codes and break codes. Were paged and enable full-text search.
• Plotting the original position when you enter the queue and there are new graphics.
• Wildcards are available for the names of queues: wildcards to select all queues or members of a queue.
• Schedule adherence: track - in the call distribution graph - how many different agents were on the phone during a timeslot.
• The duration of peak and average duration is now displayed in the call distribution graph.
• The detailed configuration of the agents can see the set of queues where each agent is a member.
• E 'can now effetutare listening live or incoming calls than outgoing, using different sections of the dialplan.
• And 'possible to have code "invisible", ie code that can not be selected from the main page but used by the system of wildcards.

In short, a great product that continues to grow in functionality and features, more oriented to a field of advanced call center.

Links:

QueueMetrics

Tags: Call Center Systems | VoIP PBX | Asterisk Consultant Naples | PBX | Voip | Asterisk CTI | PBX | IP Phones | Networking | Linux

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Development of IVR systems, callcenter, PBX Voip.

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Asterisk 1.4.15 and Asterisk 1.2.22 - The Asterisk development team has released two versions which solve the following two security holes:

• http://downloads.digium.com/pub/asa/AST-2007-025.pdf - SQL Injection vulnerability found in the form res_config_pgsql. The default installations of Asterisk are not affected by this vulnerability. However, systems using the Postgres Realtime Engine may be attacked remotely. Furthermore, this vulnerability only affects systems 1.4.x since the postgres module was introduced from version 1.4.x.
• http://downloads.digium.com/pub/asa/AST-2007-026.pdf - Other vulnerabilities like SQL Injection. The input for the ANI and DNIS fields are not handled properly. The default installations of Asterisk are not affected by this vulnerability. However, systems that use the Postgres CDR logging module module could be attacked remotely. This vulnerability affects versions 1.2 and 1.4 of Asterisk.

Asterisk-addons version 1.4.5-This version contains a few bug fixes since the previous release but it was necessary to ensure compatibility with the latest version of Asterisk, 1.4.15.

Zaptel zaptel 1.2.22 and 1.4.7 - both releases contain fixes for many drivers TC400B, a bug fix for the driver for card users wctdm24xxp WPM150M and numerous improvements and fixes to the Xorcom suite of drivers. Asterisk.org The development team has released Asterisk versions 1.4.15 and 1.2.25.

Links:

• Asterisk and Asterisk Addons: http://downloads.digium.com/pub/asterisk/
• Zaptel: http://downloads.digium.com/pub/zaptel/

Tags: Call Center Systems | VoIP PBX | Asterisk Consultant Naples | PBX | Voip | Asterisk CTI | PBX | IP Phones | Networking | Linux

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Development of IVR systems, callcenter, PBX Voip.

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