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This chapter focuses on the advanced settings for RTP (Real-time Transport Protocol) as well as the global settings for SIP, IAX and analog. Furthermore, you learn how to create Asterisk Manager Interface Accounts and customize the web interface to individual needs.
The Real-time Transport Protocol (RTP) defines a standardized format for transporting audio and video over IP networks. By default RTP uses the port range between 10000 and 10200. For some routers and firewalls it might be necessary to adjust the port range.
Another reason for adjusting the port range might be a high number of parallel calls. Each call uses two RTP ports. This means with 200 ports, 100 parallel calls are possible. If your telephone system needs to process more calls at the same time, you need to enlarge the port range.
Click Save to finish the configuration.
The Session Initiation Protocol (SIP) is a signaling protocol used by most VoIP phones.
You might want to change the SIP Binding Port (by default port 5060) to increase security. Also some SIP providers need additional parameters for Registration Timeouts and DNS Service Records.
Some firewalls close ports after periods of inactivity. This behavior might require you to reduce the registration timeouts for SIP providers. Another reason might be the need for different registration timeouts required by some SIP providers. Default values for minimum are 60, for maximum 3600 and for standard 120 seconds.
In some cases DNS Service Records might increase the time needed for the call set-up. If you use a SIP telephone but no SIP provider it may make sense to disable DNS Service Records but it is not recommended.
Manual Attributes, which are defined in Advanced/SIP, are applied for all SIP telephones and providers used with AskoziaPBX. You can find further information in Manual Attributes.
Click Save to finish the configuration.
The Inter-Asterisk eXchange protocol is a native Asterisk protocol and supported by a number of VoIP phones.
The default Binding Port for IAX (port 4569) can be changed in case it is already used otherwise or to improve security. Manual Attributes which are defined in Advanced/IAX are applied for all IAX telephones and providers used with AskoziaPBX. Manual attributes are described in Manual Attributes.
The second part of the menu is about the configuration of the Jitterbuffer. A jitterbuffer compensates time-of-arrival differences. When using a jitterbuffer, less packets need to be discarded due to late arrival (packet loss rate). On the other hand, delay is increased. Consequently, a jitterbuffer is a compromise between delay and packet loss.
You can choose between two options. Enable Jitterbuffer on IAX connections terminated by the system and Use Jitterbuffer even when bridging two endpoints. If you want to use the second option both options need to be checked. It needs to be "forced" because the control of the jitterbuffer is usually done by the telephones.
Click Save to finish the configuration.
Some phones (e.g. Snom, Polycom etc.) allow easy firmware updates via TFTP. To use AskoziaPBX's built-in TFTP server, you need to add an external storage first. Once you have mounted the storage, you can put the file(s) you'd like to provide via TFTP on /storage/usbdisk1/askoziapbx/tftpd/.
The phones usually ask for an IP address where they can find the update. Specify the IP address in Listen on IP.
In case you use different indication Tone Zones, their loading can be accelerated. If you, for example, mostly use the British tone scheme but exclusively for one telephone the French tone scheme you can prioritize the schemes using drag-and-drop.
If you only use one tone scheme just load this one.
|Select all indication tone zones to be supported. The first tone zone will be the default tone zone.|
The second configuration point affects the Port's impedance. It should be configured the way used in the country of assignment to make sure incoming call signals are interpreted correctly.
The Asterisk Manager Interface (AMI) provides the ability to communicate directly with Asterisk. This might be helpful for troubleshooting because you can monitor all events Asterisk produces internally. You can define users with different authorizations (read and write permissions) for different event types. You can monitor these events by using a Telnet client.
Telnet (Telecommunication Network) is a widely used network protocol. All common operating systems come with Telnet clients.
To add a new user account you need to specify an Username and a Secret (a password in Asterisk terminology).
Specify a Permit Network and an optional Deny Network. In Deny Network, enter the IP address and subnet mask of your network instead of the "x". Grant permissions by activating the checkboxes.
|On servers with public access you should only use the AMI when using a firewall or a SSH tunnel.|
Afterwards you need to restart your telephone system.
You can now connect to the Manager Interface via port 5038. To do so, enter the following command in your terminal (how to open the terminal in your operating system is described in Installation). Replace the "x" with the IP address of your telephone system.
telnet xxx.xx.xx.xxx 5038
Your terminal should look like this:
macbook:~ myMac$ telnet xxx.xx.xx.xxx 5038 Trying xxx.xx.xx.xxx... Connected to xxx.xx.xx.xxx. Escape character is '^]'. Asterisk Call Manager/1.1
Now you have to log in with your user information specified earlier in the AskoziaPBX web interface. Let us assume that you used the word "listener" for Username and Secret. Please note that the complete text of the next command needs to be entered. For line breaks press enter.
Action: Login Username: listener Secret: listener
A successful log in looks like this.
Response: Success ActionID: 1 Message: Authentication accepted
All permitted events can now be monitored.
AstManProxy is a multi-threaded proxy server for Asterisk. It is designed to handle communication with multiple Asterisk servers and to act as a single point of contact for applications. AstmanProxy supports multiple input and output formats, including XML, CSV, HTTP, HTTPS, SSL and the Asterisk-native standard format.
AstmanProxy is able to manage multiple requests and process them one by one. Users only see events they are related to.
More information can be found at www.voip-info.org.
For using AstManProxy, first activate the Activate AstManProxy checkbox and click Save.
Enter an Username and a Secret (a password in Asterisk terminology). Define the Channel and the context for incoming and outcoming events. Click on Save to finish your configuration.
Humbug Telecom Labs offers Telecom Analytics and Fraud Detection. Sign up for an account at the Humbug web pages.
Click on Humbug Analytics in the menu bar. Enable Humbug and enter your API Key and your Encryption Key.
|Please note that Humbug is third party software. As part of the Humbug analytics, your call details are transfered to Humbug Labs.|
Click Save to finish the configuration and reboot your telephone system.
After changing settings, click Save to finish the configuration.
The Console Menu can be disabled to prevent third parties from accessing the AskoziaPBX console via VGA or serial interface. To access the console, AskoziaPBX needs to have access to one of these interfaces. Additionally, physical access to the phone system is required.
The Navigation menu is meant to simplify the web interface by optionally hiding the SIP, IAX, ISDN and analog menus. This can be useful if your PBX does not provide interfaces for one or more of these technologies.
Furthermore, it's possible to hide the SIP provider template drop down menu. This can improve usability and clarity of the web interface if no listed provider is used.
At CDRs, you can choose to hide the last three digits of the target caller id in call detail record PDF files to enable anonymous analysis.
You can choose the Plattform Icon to be displayed in your system’s dashboard.
- Generic PC for PC-based solutions
- Embedded, for example for Desktop Telephony Servers
- Rack, for example for 19” Telephony Servers
- Virtual Machine for installations on virtual machines (VM).
To record all phone calls, enable global Call Recording. This extends the menu by the options listed below. Alternatively, you can activate call recording for single phones.
- Internal activates call recording for all internal calls.
- Inbound activates call recording for all incoming calls.
- Outbound activates call recording for all outgoing calls.
Automatic call recording permanently requires big amounts of memory (approx. 1MB/minute). The administrator of the system is responsible for providing enough memory and creating backups. Askozia is not liable for any data loss.
Automatic call recording requires a lot of CPU power, especially when doing multiple recordings at the same time. Before activating automated call recording, make sure that your hardware is sufficiently dimensioned. Otherwise, bad audio quality and dropped calls may occur.
Activating the FTP server option provides FTP access to the call recording archive in addition to the web interface access. This option is recommended for downloading large archives.
By default, an announcement will be played if call recording is activated during a phone call, such as by pressing the record button of a Snom phone, or *1 for phones from other manufacturers. This announcement can be deactivated under Disable Activation Feedback.
It follows a list of possible scenarios for automatic call recording.
When using automatic call recording, call transfers should always be initiated with the transfer button of the phone. Transfers with ** and ## are not supported in combination with call recording.
When using automatic call recording, we recommend to avoid multiple transfers of the same call or transfers out of a call flow as recordings may get lost.
Supported call recording scenarios for internal calls
- Direct calls between two internal phones
- Direct calls between two internal phones with a transfer to a third internal phone
- Calls to external phones
- Calls to internal phones, whereas one phone has call forwarding activated
Supported call recording scenarios for inbound calls
- Calls which are routed through a provider directly to an internal phone
- Calls which are routed through a provider directly to a group of internal phones (call group, hunt group)
- Calls which are routed from a provider directly to a call flow
- Calls which are routed from a provider directly to an internal phone, from where they are forwarded to another internal phone (by using the transfer function of the phone itself, or the AskoziaPBX application 000023 or 000023*xxx*yyy)
Supported call recordings scenarios for outbound calls
- Calls which are routed through a provider directly to an external number
Recorded calls can either be managed in the Call Recording Archive or via FTP-access.
Manage Call Recordings via FTP
To enable FTP access to the call recording archive, a FTP client (such as Cyberduck, FileZilla or Krusader) is required. The computer you are using to access the archive through the FTP client, needs to be in the same network as AskoziaPBX. Use Askozia’s IP address and port 21 to access the system.
- IP: IP-address-of-AskoziaPBX:21
- user name’’’: root
- password: password of the web interface
Under Virtualization, support for virtual environments of VMware and HyperV can be activated. This makes it possible for a system being shut down properly through the interface of the virtual machine (VM). Otherwise, the VM could just be switched off. In other words, you don’t need to access the Askozia web interface to shut it down properly.
Additionally, the VM also knows the IP address of Askozia and may be able to book additional memory if necessary. In general, this support improves the interaction of Askozia with its host system, and consequently the work performance.
The support of beroNet network interface cards (NIC) enables display of network cards with beroNet network address in the network settings. This option is especially important if AskoziaPBX is used on a beroNet Appliance in order to identify the MAC address of beroNet interface cards and NICs.
Bridging remaining network interfaces into a bridged interface group can be useful to establish a separate network for IP phones or to securely connect an analog or digital gateway.
Usually, Askozia logos are automatically transferred onto phones during auto-configuration. This function can be disabled by activating Hide logos.
Block all provider per default blocks all provider for new created phones. Block all provider blocks all provider for all existing phone accounts. Disable Call Flow Cache disables the caching of Call Flows while editing them.
Disabling the Call Flow caching is only recommended for powerful machines.
Disable Voicemail Instructions disables the system announcement:"Please leave your message after the tone. When done, hang up or...".
Even if voicemail instructions are disabled, the systems busy or n/a announcement is played as long as no personal announcement is uploaded. System announcement sounds like: "The user with number XYZ is not available... "
Disable call info pickup notification deactivates the information, displayed on the phone, that another member of your pick up group, picked up an incoming call for you. Allow external forwarding makes it possible to directly forward to external. If this checkbox is inactive, it's only possible to forward to internal extensions. Activate MWI polling activates a polling for the "Message Waiting Indication" which is needed by some phones to set the correct status of the voicemail box. Restart Asterisk causes a soft reboot every night. That means, Askozia will wait till no calls are active and than reboot Asterisk. With Disable ACPI the "Advanced Configuration and Power Interface" is deactivated. This optimizes CPU-Power.
Virtual Fax: Outgoing Configuration
Check for new messages allows to specify in seconds, how often AskoziaPBX should check if there are new faxes to be sent out.