About sipxecs and others

When talking about open source IP-PBX, people familiar with this field probably know AsteriskThis famous open-source ippbx. There are many technical documents about asterisk. However, what I want to introduce here is another ippbx solution, which is also an open-source system- Sipxecs. I checked Chinese information and introduction about sipxecs through Google and found that there were not many, obviously it was no longer popular with asterisk. However, asterisk is definitely better than sipxecs, and does sipxecs certainly have no advantages compared with asterisk?

1. What is sipxecs and what can it do?

According to the information on its official website (http://www.sipfoundry.org/), sipxecs enterprise Communication System (ECS) is a highly scalable, enterprise-level communication solution. It is an independent product of a non-profit, open-source organization named SIPfoundry. With standard and open-source environments, sipxecs provides low-cost, easy-to-use, and interoperability, functionality, and scalability that cannot be found in other systems.

Sipxecs is based on the SIP protocol. It provides all the typical functions of the expected PBX, including voicemail, unified message, auto-attendant, and conferencing), attendance, and call center applications.

Sipxecs is not only an Instruction Set network telephone exchange platform, but also an overall solution ). That is to say, it already contains an application component, such as web-ui, that must be put online and used by an online telephone exchange system. Sipxecs is currently the only open source IP-PBX can do terminal phone settings, can be Plug and Play System. This feature is very useful for the extensive deployment of applications in the office. However, it should be noted that, unlike asterisk, sipxecs is authorized by L-GPL software, which is roughly the same as GPL authorization, but the difference is that the function library has special authorization terms.

To sum up, sipxecs has the following features:

• Voice mail (Voice Mail): Integrated Voice Mail supports automatic response of each user's Personality
• Automatic Call Distribution System (automatic call distribution): the Integrated Call Center solution allocates multiple agents and queues for calls through intelligent routing.
• Unified Messaging: voice mail messages can be retrieved through a Web browser or forwarded to any email client.
• Multiple automatic responses (multiple auto attendants): You can easily configure automatic responses on the browser interface.
• Configuration Management: real plug-and-play. The dial-up plan, users, and terminals can be centrally controlled and managed through an intuitive browser interface.
• User Portal: a powerful web user portal allows users to separately manage key features such as time-based find-me/follow-me.
• Easy to install and use: it takes only a few hours to deploy sipxecs. At the same time, sipxecs is designed as an administrator for everyone. After the installation, you can use a web interface to complete all the operations such as adding, migrating, and changing.

Unlike asterisk, sipxecs is connected to PSTN through an external gateway, that is, through a traditional analog Relay (fxo Voice Gateway) or digital relay (t1/T1/PRI. In addition, sipxecs can also use the SIP relay provided by a service provider (ITSP) to connect to the outside. With four analog relay lines, you can use sipxecs to build an enterprise IP voice solution that supports 4 to 12 users, or use one or more digital relay lines, this allows the system to support hundreds of users. Because the sipxecs system supports distributed deployment, sipxecs IP-PBX is very scalable. For example, for an organization like a group-branch, different configurations can be performed in the branch:

(1) The Branch runs its own sipxecs instance;

(2) The branch uses the Group IP network to connect to the sipxecs for centralized management and operation;

(3) Each branch can have a redundant or non-redundant sipxecs configuration.

Local gateways can be configured for each branch. This can improve the handling of emergency calls, provide a minimum cost route, or enable elasticity to directly unload group WAN network connection calls to the local telephone network.

1. What is sipxecs and what can it do?

According to the information on its official website (http://www.sipfoundry.org/), sipxecs enterprise Communication System (ECS) is a highly scalable, enterprise-level communication solution. It is an independent product of a non-profit, open-source organization named SIPfoundry. With standard and open-source environments, sipxecs provides low-cost, easy-to-use, and interoperability, functionality, and scalability that cannot be found in other systems.

Sipxecs is based on the SIP protocol. It provides all the typical functions of the expected PBX, including voicemail, unified message, auto-attendant, and conferencing), attendance, and call center applications.

Sipxecs is not only an Instruction Set network telephone exchange platform, but also an overall solution ). That is to say, it already contains an application component, such as web-ui, that must be put online and used by an online telephone exchange system. Sipxecs is currently the only open source IP-PBX can do terminal phone settings, can be Plug and Play System. This feature is very useful for the extensive deployment of applications in the office. However, it should be noted that, unlike asterisk, sipxecs is authorized by L-GPL software, which is roughly the same as GPL authorization, but the difference is that the function library has special authorization terms.

To sum up, sipxecs has the following features:

• Voice mail (Voice Mail): Integrated Voice Mail supports automatic response of each user's Personality
• Automatic Call Distribution System (automatic call distribution): the Integrated Call Center solution allocates multiple agents and queues for calls through intelligent routing.
• Unified Messaging: voice mail messages can be retrieved through a Web browser or forwarded to any email client.
• Multiple automatic responses (multiple auto attendants): You can easily configure automatic responses on the browser interface.
• Configuration Management: real plug-and-play. The dial-up plan, users, and terminals can be centrally controlled and managed through an intuitive browser interface.
• User Portal: a powerful web user portal allows users to separately manage key features such as time-based find-me/follow-me.
• Easy to install and use: it takes only a few hours to deploy sipxecs. At the same time, sipxecs is designed as an administrator for everyone. After the installation, you can use a web interface to complete all the operations such as adding, migrating, and changing.

Unlike asterisk, sipxecs is connected to PSTN through an external gateway, that is, through a traditional analog Relay (fxo Voice Gateway) or digital relay (t1/T1/PRI. In addition, sipxecs can also use the SIP relay provided by a service provider (ITSP) to connect to the outside. With four analog relay lines, you can use sipxecs to build an enterprise IP voice solution that supports 4 to 12 users, or use one or more digital relay lines, this allows the system to support hundreds of users. Because the sipxecs system supports distributed deployment, sipxecs IP-PBX is very scalable. For example, for an organization like a group-branch, different configurations can be performed in the branch:

(1) The Branch runs its own sipxecs instance;

(2) The branch uses the Group IP network to connect to the sipxecs for centralized management and operation;

(3) Each branch can have a redundant or non-redundant sipxecs configuration.

Local gateways can be configured for each branch. This can improve the handling of emergency calls, provide a minimum cost route, or enable elasticity to directly unload group WAN network connection calls to the local telephone network.