As you know Digium is releasing Asterisk versions very quickly. Recently Asterisk Major release introduced in the telephony playground and that is Asterisk 1.8.0. This is basically under beta but still many telephony companies are concerned with this version as this will be lasting for 4 years and there will be no major release in this four years time. Also users going to work on this in their labs because it has some outstanding features.

If I talk about my expereince with Asterisk 1.8.0 testing, here you go:

I have installed 1.8 on my development machine for testing purposes. From the look and feel did not really know what changed.

But beneath the surface a lot is happening for me was the importance of MGCP / NCS support. Also there're some very features added in SIP and Queues applications. Also alot of other applications and dialplan functioned added in this version. The current status beta2 gibts there but a few mistakes, even one caused a crash of the Asterisk, which is a patch from the Asterisk ISSUE tracker can clean up but by the error in the beta2, which has led to a crash in from the Beta 3 fixed, although there are a few bugs in all beta versions.

Here is brief overview:

• IPv6 support for some VoIP protocols (I have not yet tested)
• MGCP / NCS for the connection of VoIP cable modems
• Full support G.722
• Full T.38 support
• and much more …

Give it a try by yourself in your lab. Test it thoroughly from all aspects. But make sure you don't put Asterisk 1.8 in production at this time. I'd suggest to give it a couple of months and let it to be stable but meanwhile you can study it from your relevant VoIP protocols, applications and other functions. Soon, I'll share the lis of Asterisk 1.8 features. If you have any questions or comments, please give your feedback on Asterisk 1.8 in comments area.

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This guide will demonstrate the process of configuring a Linksys PAP2 ATA with Asterisk to make calls between two regular phones. Operating system used in this tutorial is Debian GNU / Linux (etch), kernel version 2.6.18-5-486 was used. 

Environment used 

- ATA Linksys PAP2; 

- 2 analog phones; 

- A hub; 

Requirements 

- Asterisk installed and configured correctly 

Installing DHCP 

First you must take care not to put two DHCP servers on the same network, it can be used for a hub to link the local DHCP and ATA. To install the dhcp server will use the APT package manager, as illustrated in the following command. 

# Apt-get install dhcp3-server 

DHCP Configuration 

To configure the DHCP service is needed to edit / etc/dhcp3/dhcpd.conf. 

option domain-name "voffice.com.br"; 

option domain-name-servers 192.168.1.1; 

option subnet-mask 255.255.255.0; 

default-lease-time 600; 

max-lease-time 7200; 

server-name "VOffice"; 

subnet 192.168.1.0 netmask 255.255.255.0 ( 

range 192.168.1.100 192.168.1.150; 

ATA host ( 

hardware ethernet 00:12:17: FC: 36: AB; 

fixed-address 192.168.1.135; 

) 

) 

It can be observed in the above configuration, the IP address of the Linksys ATA has been configured according to their MAC address. 

Configuring the Linksys PAP2 ATA 

To configure the two lines of the Linksys ATA is necessary to access it using any browser with an IP address set to the same. 

http://IP-DO-ATA/admin/advanced 

LINE 1 

- Access the menu option "Line 1" 

- Line Enable: yes 

- Proxy: 192.168.1.1 

- Register: yes 

- Display name: 1000 

- Password: 1000 

- User ID: 1000 

LINE 2 

- Access the menu option "Line 2" 

- Line Enable: yes 

- Proxy: 192.168.1.1 

- Register: yes 

- Display name: 1001 

- Password: 1001 

- User ID: 1001 

The above options may vary according to the needs of each. Finished the setting, click Save Settings. 

Adding the extensions on Asterisk 

To add the extensions in Asterisk you must edit the file / etc / asterisk / sip.conf and add the configuration shown below. 

# Vi / etc / asterisk / sip.conf 

[1000] 

type = friend 

username = 1000 

callerid = 1000 

secret = 1000 

host = dynamic 

context = test 

[1001] 

type = friend 

username = 1001 

callerid = 1001 

secret = 1001 

host = dynamic 

context = test 

Adding a context in Asterisk 

Now you must create the context test, the phones can make calls, so you must edit / etc / asterisk / extenions.conf. 

# Vi / etc / asterisk / extensions.conf 

[Test] 

exten => _XXXX, 1, Dial (SIP / $ (Extensible)) 

exten => _XXXX, Hangup 

Final Thoughts 

After carrying out all the settings correctly you must start the asterisk, as illustrated below. 

# Asterisk-r 

VOffice * CLI> sip reload 

VOffice * CLI> extensions reload 

VOffice * CLI> sip show peers 

Name / username Host Dyn Nat ACL Port Status 

1001/1001 192.168.1.135 D 5060 Unmonitored 

1000/1000 192.168.1.135 D 5060 Unmonitored 

Now test calls can be performed from one analogue phone to other one.

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For some time, Cisco was interested in the compatibility of its VoIP telephony related products with other VoIP vendors, very specifically, to bring their handsets compatible with open-source PBX systems, and more specifically to Asterisk and manufacturers developing their products especially oriented to this system, Digium.

But this is not the only vendor, 3Com has been talking with Digium, Asterisk hardware manufacturer, to use appliances such as communications solution itself.

This statement of intent, rumor mill at the moment, Cisco clashes with the launch of the new range of IP telephony terminals 6900 Series.

This range of terminals using the communication protocol SCCP, Cisco proprietary protocol. Do not use, nor has any support for SIP protocol, mainly used by other manufacturers, in particular communication systems based on Asterisk.

As we know Cisco, Linksys in its range, has terminals compatible with other phone systems.

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Below are listed some features of this product:

• Connect to Skype users that are currently online

• Call using SkypeOut

• Make a connection with SIP channels

• Make several simultaneous connections (limited only by hardware)

More information: http://www.chanskype.com

Source: ChanSkype

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Since Asterisk is most famous opensource telephony platforum out there. The main reason of the popularity is its integration various telecoms service through telephony modules and hardware. Hence most of the people confuse when they upgrade their Asterisk and they do not find chan_zap module in the Asterisk modules directory. Asterisk wonder what’s wrong with the hardware. They do not know that Digium has changed the zaptel package name with Dahdi. I was reported a lot of questions about Zap and Dahdi difference. Below is a little explain that what is DAHDI? what’s difference between Zaptel and Dahdi and how it will effect your Asterisk infrastructure in future.

DAHDI (Digium Asterisk Hardware Device Interface). But it seems that Astricon’s joke was not entirely a joke: Digium Zaptel development stops and rename it DAHDI (Digium Asterisk Hardware Device Interface). Approximately two years ago, the owner of the trademark has contacted the company and Digium Zaptel advised that the name is actually a brand name. The company produces a phone card and would like to see on demand ‘phone card Zaptel’ not granted would not be associated with their brand name products.

In connection with the situation and to prevent it in the future, the new name will be registered as a brand (like Asterisk). For these purposes, will be released version DAHDI 2.0.0, which would include virtually all the functions of Zaptel 1.4 (except for supporting kernel 2.4 and drivers for older cards torisa and wcusb).

Asterisk 1.6 will be able to use only DAHDI, previous versions will remain compatible with Zaptel 1.4 (Asterisk 1.4 cmozhet used as Zaptel, and DAHDI). After version DAHDI 2.0.0 Issue Zaptel with correction of errors will stop.

The file is replaced by zaptel.conf /etc/dahdi/system.conf and although they are virtually equal on the inside, there is a small detail, the echo canceler that can dynamically load and unload and select independently for each DAHDI channel, and if we have a primary E1 (30 channels of voice) can use an echo canceller for the first 10 channels, another echo canceler for the 10 following another for the last 10. ¿Useful? it is not now, but sure someone will find it. For everything else, is exactly the same.

Moreover, the file has become zapata.conf /etc/asterisk/chan_dahdi.conf and true, is almost the same inside, the differences are minimal and some quite odd that I leave you to discover

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Digium, a company that promotes Asterisk in the corporate segment, is preparing for the integration of open PBX infrastructure providers of telecommunications services. A partnership agreement reached with the British manufacturer of telecommunications equipment Integrics.

The deal could serve as a starting point for Digium, sponsoring Asterisk, which sees the Integrics and its product Enswitch a good basis for further joint development of market VoIP. Currently, this niche is firmly hold manufacturers propreitarnogo equipment.

With the help of Digium Integrics expects to expand its user base, which currently has about 25 companies, which deployed 40 PBX. Among them is the largest company providing IP telephony services to more than one hundred thousand of its customers and thousands of simultaneous calls.

Digium is not just a marketing support Asterisk, but he produces equipment. Although, it must be noted that while the equipment is suitable only for small businesses. For example, AA350 Switchvox is no more than 400 users, AA250, which was declared as the ability to serve up to 500 users, never went beyond the laboratory.Osnovnoy Digium aim is to help communications service providers to gain profits from the promotion of open solutions segment of small and medium-sized businesses.

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SIP Changes:

One of the most significant change is that now SIP signaling can be carried out not only on the UDP protocol, but also for TCP.
The mere use TCP instead of UDP does not provide many benefits, but allows the use of TLS encryption bandwidth that is claimed by companies that are making higher demands on security.
It is worth noting as well the possibility of changing the timing for the SIP protocol, the so-called timers, defined in RFC 4028. This may require, for example, in cases where the network is characterized by stable long delay packages, as well as avoiding the hang SIP sessions, when communication with the SIP client fetches up during the session itself.
Less substantial but facing mention changes in the SIP steke:
- Ability to specify the port number for bindaddr, externip and externhost parameters.
- Options “musiconhold” and “musicclass” in the file sip.conf no longer exists, they are now replacing “mohsuggest” and “mohinterpret”.
- Function SIPPEER gives the port number and call pickup groups.
- Restructuring Mechanism MWI (Message Waiting Indication – notification of new voice messages) led to the scrapping Options checkwmi file sip.conf and adding pollmailboxes in sip.conf and pollfreq in voicemail.conf.
- Added the ability to send text in real-time mode on a protocol T.140. Who does not know what it is – see http://en.wikipedia.org/wiki/Text_over_IP.
- During the registration of SIP clients, the answer is 100 Trying default will no longer be sent as According to RFC is not required. For inclusion, it is necessary to clearly specify registertrying.
- To better accountability needed for queues and SIP subscriptions added a new option callcounter. Earlier this requires was the option call-limit to deliberately inflated value.
- Added option qualifyfreq, which sets the frequency of inspections qualify.

IAX2 Changes:

- Added new option trunkmaxsize, regulating the maximum number of channels in IAX2 Trunk.
- Added option srvlookup allowing use DNS SRV records to determine the IP address. In SIP steke this functionality was implemented much earlier.
- Added support for OSP – «http://www.transnexus.com/OSP Toolkit / what_is_osp.htm».

Changes in Local Channel:

- Function DEVICE_STATE () indicates INUSE or NOT_INUSE for existing ekstenshenov.
- Ability to incorporate Jitter buffer for Local channels, working in conjunction with a team option j Dial and the suffix / n.

Change the channel DAHDI (formerly Zap):

Zap channel was renamed the DUHDI http://blogs.digium.com/2008/05/19/zaptel-project-being-renamed-to-dahdi/.

Changes in 1.6. include:
- Support for SS7 protocol, using the library libss7 (yes neuzhto!).
- Team dialplan show now displays information coincidentally c CID (number of caller), which can be used in dialplane after ekstenshena through direct slash.
- New command CLI: dahdi show version.
- In the configuration file chan_dahdi.conf an option setvar.
- You can now include notification of new messages (MWI) for FXO lines through the options and mwimonitor mwimonitornotify.
- Added the type of alarm auto, with its use of Asterisk will try to use the alarm for the channel, which is listed in DAHDI.
- New command dahdi set dnd from the command line interface allows you to change the Do-Not-Disturb status for the channel. In this event DNDState Manager (AMI) is generated irrespective of how it changed DND state.

Changes in Dialplan functions:

- Added option DEVICE_STATE () to obtain information on the status of various devices. The function can also be used to establish their own state and control of dialplana.
- Team MailboxExists turned into a feature.
- A new option for the Dial, which IP phones do not count the call as missing in the case of non, or — the lifting of the Asterisk.
- Added option HINT (), provides information stored in dialplane in a special alphanumeric prioretete hint.
- Added option SYSINFO () to obtain information related to the operating system.
- Added features TOUPPER () and TOLOWER () name spoken for themselves and are well known to programmers.
- When connecting channels, Asterisk sets variable BRIDGEPVTCALLID equally unique ID call driver channel
(if the driver has it). For SIP calls, for example, the variable will be field SIP call-ID.

It should be noted that despite the addition of new capabilities and the emergence of more convenient methods for implementing the desired functionality, Digium developers decided not to remove obsolete functionality of dialplan to preserve backwards compatibility.

Changes in command line interface (CLI):

In Asterisk 1.6.0 appeared commands such as:
- Sore show hint.
- Core show settings.
- Core show channels count.
- Queue pause member and queue unpause member.
- Possibility to change the parameters ulimit without restarting Asterisk.
- When you switch agi debug before the conclusion of AGI message will name the channel that facilitates debugging
on the operating systems dialplan set extenpatternmatching true / false.
- Now you can be placed variable channel from the CLI commands through a core set chanvar.
- Added Section startup_commands in cli.conf, where you can specify command of the CLI, performed immediately after running Asterisk.
- Team devstate change allows a CLI done the same thing as a function DEVICE_STATE () in dialplane.

Changes in voice mail (VoiceMail):

- Now you can customization of what sounds to use for specific items on the voice menu.
- voicemail show users shows users configured realtime basis.
- MWI algorithm has been changed. Now it is not based on periodic inspection, and to generate events. Different modules can subscribe to MWI and Asterisk will inform them about changes in golosvyh boxes. This also means that if change the contents of the voice mail from outside, such as web intferfeysom to voice mail, Asterisk still have to be Periodically check the contents of voice mailboxes. To do this, set up options pollmailboxes and pollfreq.
- Added the ability to store greetings to the IMAP server.
- Ability to change the buttons for scrolling, pause and stop playing a voice message.
- Added option lockmode in asterisk.conf to specify the method of blocking file voice mail and other files in the future. By default method of creating lockfile. Now there is another method – flock, which can work on file systems — SMB / CIFS.
- Added the ability backup / delete messages it for the user to restore accidentally deleted message.

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• Pay the bond in the amount of the total cost;
• Pay for delivery (not required for own);
• Use the equipment properly;
• Return the equipment in up to 10 days of receipt;
• To recover money deposit (after checking maps).

In the event that the equipment is not returned on time, it is considered to be sold.

In all other cases, we take the equipment back without charging a fee for use.
Note: The guarantee deposit paid in cash.

Order equipment for testing

To make an order for testing equipment, use a standard form to send an application.

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The following list describes the functionality as the Contact Center to carry out his task:

• Production calls in the queue and intelligent distribution of the operators contribute to the efficient processing of incoming and outgoing calls;

• Receive messages by e-mail. WEB-mail and via the company allows customers to do business with the company, as it is convenient;

• Information services to customers using an interactive voice menu allows customers to obtain their own information they need, or select their unit or staff member;

• Automatic outbound call customers for new services (active sales), customer information;

• Inxtegratsiya closely with internal and external business processes to improve the efficiency of handling requests for services and products for handling complaints and wishes of customers;

• Individual approach to each client by identifying the history of previous calls and receive call him a friend operator;

• Analysis and development of new services and products through the analysis of customer referrals;Automatic organization of repeated calls offers pending sale of a product or service.

More details here:

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