Gateways in sipXecs 4.0: Part 2

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by Michael W. Picher | August 2009 | Networking & Telephony

Read Gateways in sipXecs 4.0: Part 1 here.

Advanced Parameters

The Advanced Parameters settings, shown as follows, are accessed by clicking on the Advanced Parameters item in the lefthand menu and are a collection of AudiCodes-specific settings.

Gateways in sipXecs 4.0: Part 2

The following configuration options are available on this page (click on Show Advanced Settings to reveal all options):

  • Secure SIP Calls: If this is enabled, gateways will only accept SIP calls from IP addresses listed below. The default is disabled (unchecked)
  • Accepted IP Addresses: They are used in conjunction with the above setting. It is a space-separated list of IP addresses from which gateway will accept the calls. It is taken into account only if Secure SIP Calls is enabled. The default is the IP address provided for the PBX during install.
  • Digit Delivery to Telephony Port: This setting enables a digit string to be played to the port at the far end, after off-hook. The default is disabled (unchecked).
  • Digit Delivery to IP: This setting enables a digit string to be played to the port at the far end, after off-hook. The default is disabled (unchecked).
  • DID Wink Support: When this is enabled, the gateway can connect to EIA/TIA 464B Loop Start DID lines. Both generation and detection are supported. The default setting is disabled (unchecked).
  • Enable Call Disconnect on Polarity Reversal: If this is checked, enables port disconnect (on-hook) based on polarity reversal. Some POTS providers will reverse the polarity of the analog phone line to signal disconnection to a PBX. The default is disabled (unchecked). Enable Call Disconnect on Current Drop: If this is set to 1, enables port disconnect (on-hook) based on current drop. The default setting is disabled (0).
  • Enable Call Disconnect on Broken Connection: If this is checked, the call is released if the gateway stops receiving RTP for a period of time. The default is enabled (checked).
  • Broken Connection Timeout (10msec): The amount of time for which RTP is not received, before the call is cleared. In 10 ms steps, the default is 500, 10 ms steps (5 seconds).
  • Enable Call Disconnect on Far End Silence: If this checked, enables disconnection of call based on silence. The default is disabled (unchecked).
  • Silence Period for Disconnect: The detection period, in seconds, before the call is released based on silence. The default is 120 seconds.
  • Silence Detection Method: This setting can be set to "None" (silence detection option is disabled), "Packets Count" (according to packet count), "Voice/Energy Detectors" (according to energy and voice detectors (default)) or "All" (according to packet count and energy / voice detectors).
  • Silence Threshold: The threshold of packet count, in percentage, below which is considered as silence. The default setting is 8 packets.
  • Detail Level in Debug Log: The detail level of the log messages sent to syslog server. Default is 0 (off), max is 5 (full).
  • CDR Server IP Address: An optional separate syslog server to collect CDRs only. If null and CDR is enabled, the output is mixed with the log messages and is routed to the syslog server IP. (Default: 0.0.0.0)
  • CDR Report Level: This can be set to "None", (Call Detail Recording or CDR information isn't sent to the Syslog server, which is the default value), "End Call" (CDR information is sent to the Syslog server at the end of each Call) and, "Start & End Call" (CDR information is sent to the Syslog server at the start and at the end of each Call).
  • Port Busy-Out Method  : If there is a network-side failure on the gateway, the POTS interfaces can be set to busy. If this is  checked, telephony ports are busied out (special tone) in case of LAN failure or proxy communication failure. The default is disabled (unchecked).
  • Delay After Reset [sec]: Amount of time delay before answering calls after gateway reset. The default is 7 seconds.
  • Max Number of Active Calls: This is the maximum number of active calls the PBX can process. It should be set to the number of PSTN lines active.
  • Max Call Duration [min]: This is the maximum duration of a phone call. The default is 0, which means no maximum.
  • Enable LAN Watchdog: When LAN Watchdog is enabled, the gateway's overall communication integrity is checked periodically. If no communication for about 3 minutes is detected, the gateway performs a self test. If the self test succeeds, the problem is logical link down (for example, Ethernet cable disconnected on the switch side), and the "Busy Out" mechanism is activated if enabled (EnableBusyOut = 1). Lifeline is activated if enabled. If the self test fails, the gateway restarts to overcome an internal fatal communication error (default: unchecked).
  • Enable SAS: This setting is to enable/disable Stand-Alone Survivability (SAS) (default: unchecked). • SAS Registration Time: This is the time after which SAS is enabled (default: 20 seconds).
  • SAS Local SIP UDP port: This is the UDP Port for SAS SIP signaling (default: 5080).
  • SAS Local SIP TCP port: This is the TCP Port for SAS SIP signaling (default: 5080).
  • SAS Local SIP TLS port: This is the TLS Port for SAS SIP signaling (default: 5081).
  • SAS Default Gateway: This is the SAS Default Gateway IP address (default blank).
  • SAS Short Number Length: This is the SAS short number length.

Click on the Apply button to keep any changes made on this page.

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Supplementary Services

Supplementary Services are calling services that are provided by User Agents (UA). With SIP, much of the call handling is done by the intelligent end points instead of the PBX.  Clicking on the Supplemenary Servicesmenu item will display the following page:

Gateways in sipXecs 4.0: Part 2

The following configuration options are available on this page (click on Show Advanced Settings to reveal all options):

  • Enable Call Hold: This allows the gateway to enable the "Call Hold" function. The default is enabled (checked).
  • Call Hold Signaling Method: "Call Hold" can be signaled to another UA in a couple of different ways with the AudioCodes gateway. The connection IP address in SDP is 0.0.0.0 (default) or "Send Only" where the last attribute of the SDP contains 'a=sendonly'.
  • Enable Transfer: Enables the "Call Transfer" feature in the gateway. The default is enabled (checked).
  • Enable Call Waiting: Enables the "Call Waiting" feature in the gateway. The default is disabled (unchecked).
  • Hook-Flash Code: Determines a digit pattern that, when received from the Telecom side, indicates a "Hook-Flash Code". The valid range is a 25-character string. The default is blank (don't look for hook flash from the Telecom side).

Click on the Apply button to keep any changes made on this page.

FXO

The FXO settings page (shown as follows) allows some tuning of the interface with the telecommunications provider.

Gateways in sipXecs 4.0: Part 2

The following configuration options are available on this page (click on Show Advanced Settings to reveal all options):

  • Waiting for Dial Tone: When an outgoing call is made, the gateway waits to hear the dial tone on the line before dialing. The default is enabled (checked).
  • Time to Wait before Dialing [msec] : After the gateway goes off-hook to dial out, it will wait for this amount of time before dialing the phone number. The default setting is 1000 milliseconds (1 second).
  • Ring Detection Timeout [seconds]: Ring Detection Timeout is used in conjunction with inbound caller ID detection. If Caller ID is enabled, then the FXO gateway seizes the line after detection of the second ring signal (allowing detection of caller ID, sent between the first and the second rings). If the second ring signal doesn't arrive for Ring Detection Timeout, the gateway doesn't initiate a call to IP. The default setting is 8 seconds.
  • Reorder Tone Duration   [seconds]: Before releasing the line, FXO gateway plays a "Busy" or "Reorder" tone, which has duration in seconds. The default is 0 seconds disabling the reorder tone.
  • Answer Supervision: If this is enabled (checked), the FXO gateway sends a SIP 200 OK (to INVITE) messages when speech/fax/modem is detected. If it is disabled, a SIP 200 OK is sent immediately after the FXO gateway finishes dialing. The default setting is disabled (unchecked).
  • Rings before Detecting Caller ID: Determines the number of rings to wait for the caller ID from the phone company. The default is 1 ring.
  • Send Metering Message to IP: Sends a metering tone INFO message to the IP of 12/16 kHz metering pulse. Not used by sipXecs, so the default is disabled (unchecked).
  • Disconnect on Busy Tone: If the gateway detects a busy tone it will interpret this as a disconnect signal and hang up the line. The default is enabled (checked).
  • Enable Call Disconnect on Dial Tone: If the gateway detects a dial tone on the phone line, it will interpret this as a disconnect signal and hang up the line. The default is enabled (checked).
  • Guard Time Between Calls: Sometimes, after a call is ended and the gateway goes on-hook, a delay is required before placing a new call (and performing the subsequent off-hook). This is necessary to prevent the phone company thinking that the gateway is trying to perform a hook-flash. The default is 1 second.
  • First Default Caller ID   : The value that is displayed for incoming calls when no caller ID information is available. The gateway will add a port identification suffix. The default value is set to "Unknown".

Click on the Apply button to keep any changes made on this page.

Network

The Network settings page has some typical network settings. They are given as follows:

Gateways in sipXecs 4.0: Part 2

The following configuration options are available on this page (click on Show Advanced Settings to reveal all options):

  • DHCP Enabled: The default is disabled (unchecked).
  • Static NAT Traversal Enabled: Disable/enable static NAT traversal feature. The default is enabled (checked).
  • Public Address for NAT: This is the IP address of the gateway as seen from the outside of NAT. The default is no IP address (0.0.0.0).
  • Primary DNS Server: The Primary DNS server (default: 0.0.0.0).
  • Secondary DNS Server: The Secondary DNS server (default: 0.0.0.0).
  • NTP Server IP Address: The NTP Server to sync time/date with (default: pool.ntp.org).
  • NTP UTC Offset (hours): The gateway's location offset from UTC (default: 0).
  • NTP Update Interval (sec): The NTP update interval, in seconds (default: 86400 seconds).
  • STUN Enabled: Enable/disable STUN(default: unchecked).
  • Primary STUN Server IP Address: IP address of the primary STUN server (default: 0.0.0.0).
  • Secondary STUN Server IP Address: The IP address of the secondary STUN server (default: 0.0.0.0).
  • Syslog Output Enabled: Enable/disable syslog facility for diagnostics (default: unchecked).
  • Syslog Server IP address: The IP address of the syslog server (default: 0.0.0.0).
  • Syslog Server Port: The Default syslog port (default: 514).

Click on the Apply button to keep any changes made on this page.

Note that with the gateways, most of these parameters should be hard-coded into the gateway configuration rather than relying on the gateway to download the settings. Please refer to your manufacturer's documentation for detailed information as to how to do this.

Media

The Media section of the gateway configuration allows fine-grained control over the audio portion of a call through the gateway. The top of the Media settings page is shown as follows:

Gateways in sipXecs 4.0: Part 2

The following configuration options are available on this page (click on the Show Advanced Settings hyperlink to reveal all options):

  • Voice Volume: The voice gain control is given in decibels. This parameter sets the level for the transmitted Voice Volume (IP to Telecom direction) signal. The default is no gain (0). Increase this value if the far-end party is complaining of low volume.
  • PCM Input Gain Control    : The voice gain control (Tel-to-IP direction). The default is no gain (0). Increase this value, if your local callers complain about low volume with calls through the gateway. Increasing the volume too much, however, will increase static and echo will be heard on calls.
  • Silence Suppression: This suppresses transmission of silence (default: disabled).
  • Echo Cancellation: This enables echo cancellation on the gateway (default: checked).
  • DTMF Transport Options: This determines how DTMF is handled (default: Erase digit from voice stream, relay as RFC2833).
  • DTMF Volume: The DTMF gain control value in dB (to the analog side) (default: -11). Increase if DTMF is not being recognized when dialing out to another system and DTMF digits need to be sent by the caller.
  • Answer Detector Sensitivity: This helps the gateway to detect if an outbound call has been answered (default: most sensitive).
  • Fax Transport Mode: This mode shows how faxes are transported over IP (default: T.38).
  • Fax Transport Codec in By-Pass Mode: (default: G711 u-Law).
  • Fax Transport Payload ID in By-Pass Mode: Payload type for by-pass fax transport, 96 through 120 (default: 102).
  • Fax Relay Redundancy Depth: The number of times each fax relay payload is retransmitted to the network (default: 0).
  • Fax Relay Enhanced Redundancy Depth: The number of times the control packets are retransmitted when using the T.38 standard (default: 2).
  • Fax Relay ECM: Error Correction Mode or ECM is enabled or disabled (default: checked).
  • Fax Relay Max Rate: The maximum rate at which fax relay messages are transmitted (outgoing calls) (default: 14.4 kbps).
  • Fax/Modem Bypass Packing Factor: The number of (20 millisecond) coder payloads that are used to generate a fax/modem bypass packet (default: 1).
  • CNG Detector Mode: In "Relay" mode, CNG (fax tones) are detected on the originating side. CNG packets are sent to the remote side according to T.38 and the fax session is started. In "Events Only" mode, CNG is detected on the originating side. The CNG signal passes transparently to the remote side and the fax session is started. (Default: Disable)
  • Enable Caller ID: This will enable generation (FXS) and detection (FXO) of caller ID on the telephony ports (Default: checked).
  • Caller ID Display Type: (Default: Bellcore).
  • V.21 Modem Transport Type: (Default: Disable (Transparent)).
  • V.22 Modem Transport Type: (Default: Disable (Transparent)).
  • V.23 Modem Transport Type: (Default: Disable (Transparent)).
  • V.32 Modem Transport Type: (Default: Disable (Transparent)).
  • V.34 Modem Transport Type: (Default: Disable (Transparent)).

Click on the Apply button to keep any changes made on this page.

Building Enterprise Ready Telephony Systems with sipXecs 4.0 Leveraging open source VOIP for a rock-solid communications system
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RTP/RTPC

The RTP/RTPC settings allow fine-grained control over the IP side of voice conversations.

Gateways in sipXecs 4.0: Part 2

The following configuration options are available on this page (click on Show Advanced Settings to reveal all options):

  • Jitter Buffer Minimum Delay: The Jitter Buffer minimum delay, in milliseconds. The default is 10 milliseconds. Packet jitter is an average of the deviation from the network mean latency (a network that has a constant latency has no jitter). A jitter buffer helps compensate for changes in latency on the network by providing a buffer (queue).
  • Jitter Buffer Optimization Factor: AudioCodes Dynamic Jitter Buffer frame error / delay optimization factor. Valid values are 0 through 13 with the default being 10.
  • RTP Redundancy Depth: Enable or disable the generation of RFC 2198 redundancy packets (default: unchecked).
  • RFC 2198 Payload: Applicable only if RTP Redundancy Depth is enabled (default: 104).
  • RFC 3389 CN Payload: If this is enabled, SID (comfort noise) packets are sent with the RTP SID payload type according to RFC 3389. Applicable to G.711 and G.726 coders. Otherwise a proprietary method is used. The default is disabled (unchecked). Comfort noise is white noise generated by a gateway so that the person on the phone knows that a call is still active.
  • Base UDP/RTP Port: The base UDP port for RTP (default: 6000).
  • Remote RTP Base UDP Port: (Default: 0.)
  • RTP Multiplexing Local UDP Port: (Default: 0.)
  • RTP Multiplexing Remote UDP Port: (Default: 0.)
  • Comfort Noise Generation Negotiation   : The use of CN is indicated by including a payload type for CN on the media description line of the SDP. The gateway can use CN with a codec whose RTP timestamp clock rate is 8,000 Hz (G.711/G.726). The static payload type 13 is used. The use of CN is negotiated between sides; therefore, if the remote side doesn't support CN, it is not used. Note: Silence Suppression must be enabled to generate CN. (Default: unchecked.)
  • Call Progress Tones Filename: A region-specific, telephone exchange-dependent file that contains the call progress tone levels and frequencies that the VoIP gateway uses. The default CPT file is U.S.A. A different CPT file can be uploaded and will be placed in the TFTP folder and loaded during boot.
  • FXS Loop Characteristics Filename: Used for FXS (station) gateways, the name of the FXS loop characteristics definition file, to be TFTP-loaded during boot (default: MP11x-02-1-FXS_16KHZ.dat).
  • Country Variant: A definition for country variant for line characteristics. North America = 70 (default: 70).
  • LifeLine Type: On FXS gateways, a single Lifeline, connected to port #1 via a splitter (not supplied), is available. On combined FXS/FXO gateways, a splitter isn't required. All FXS ports are automatically connected to FXO ports (FXS port 0 to FXO port 4 and so forth). On FXO gateways a Lifeline isn't available. The Lifeline is activated on: 0 = power down (default), 1 = power down, or when link is down (physical disconnect), 2 = power down or when link is down, or on network failure (logical link disconnect).
    Click on the Apply    button to keep any changes made on this  page.

Management

The Management configuration settings for gateways allow some network managemet parameters to be configured for use with the network tools.

Gateways in sipXecs 4.0: Part 2

The following configuration options are available on this page (click on Show Advanced Settings to reveal all options):

  • Enable SNMP: The user can enable/disable an external SNMP manager to receive alarms/traps (default: unchecked).
  • SNMP Manager IP Address: The external SNMP Manager IP address (default: blank).
  • SNMP Port: TCP/IP default port for listening for SNMP (default: 161).
  • SNMP Trap Port: TCP/IP default trap port destination (default: 162).
  • Enable SNMP Trap Sending: Enable/disable sending traps to the external SNMP manager (default: unchecked).
  • SNMP Trusted Manager for Configuration: Enable/disable SNMP-based configuration control and trusted manager IP (default: 0.0.0.0).
  • Read-Only Community String: The SNMP management system uses the read only community string.
  • Read-Write Community String: The SNMP management system uses the read/write community string.
  • Trap Community string: The SNMP management system uses the trap community string.

Click the Apply button to keep any changes made on this page.

Unmanaged gateways

The initial configuration of unmanaged gateways looks quite similar to that of a managed gateway. There is, however, no detail that can be set because configuration files are not built for these gateways.

It is up to the system administrator to manually configure any unmanaged gateways. If there are settings that you are not sure about, refer to the Managed Gateway settings for a possible answer.

The sipXecs community Wiki has a section for common gateway configurations. The Wiki can be accessed from http://www.sipfoundry.org. Since there are only a couple of managed gateway manufacturers supported at present in sipXecs, many installations utilize unmanaged gateways. There are many gateway manufacturers with products to meet every need.

Add gateway

To add an unmanaged gateway, click on the Add new gateway drop-down box and select Unmanaged gateway. The unmanaged gateway configuration page will be displayed as follows:

Gateways in sipXecs 4.0: Part 2

The following configuration information can be configured on this page (click on Show Advanced Settings to display all configuration items):

  • Name: A name is given to the gateway (no spaces).
  • Address: The IP address of the gateway or the fully qualified hostname of the gateway (see manufacturer's documentation for information on configuring IP addressing and other basic settings).
  • Port: An optional setting for UDP or TCP port if a non-standard port is used. Set to 0 to ignore this field.
  • Transport protocol: This can be manually configured to UDP or TCP to force the SIP transport protocol. If it is set to Auto, the transport is determined through a DNS query.
  • Location: It is possible to restrict the gateway by selecting a specific location for which it can be used. A location is represented by a group of users. A user group must be created for every location that needs to be distinguished (remember that users can be in more than one group). This setting allows routing of calls based on in which location or from which user the call originates (source routing). This is useful if users located in a branch office would like to have a gateway preference so  that calls are routed through their local gateway. That is done to preserve WAN  bandwidth or to use the caller ID offered by an analog gateway based on the PSTN number assigned to it. Only if that gateway is not available, call will routing  fall back to other gateways specified for the corresponding dialing rule.
  • Shared : If this setting is checked, this gateway can be  used by any user in any location, even if a specific location is selected. This  setting is checked by default so that users in an identified location still use  their preferred gateway, but the gateway can also be used by other users in other locations.
  • Description: This is for documenting the system configuration. Information about the lines connected to the gateway is very useful here. With all of the configuration information entered, click on the OK  button and the Gateway page will be displayed as follows with the new gateway on it.

With all of the configuration information entered, click on the OK button and the Gateway page will be displayed as follows with the new gateway on it. Click on the new unmanaged gateway's name to reveal more configuration options, as shown in the following screenshot:

Gateways in sipXecs 4.0: Part 2

In the following subsections, we'll explore the unmanaged gateway settings available (luckily there aren't as many as with the managed gateways).

Caller ID

Outbound caller ID is determined by user caller ID settings combined with gateway caller ID settings. For SIP trunks, the resulting caller ID will depend on the ITSP policies. Analog gateways are typically unable to transmit caller ID and will show their respective PSTN line number instead. This is generated on the telecom provider's side.

Behavior of digital (T1/E1) gateways and SIP phones is vendor specific. Effective caller ID may also depend on your telecom provider. Advanced settings allow for overriding Display Name, domain name, and URL parameters of the caller ID. These options are typically useful only if the gateway routes calls to another SIP system

Gateways in sipXecs 4.0: Part 2

The following configuration information can be set on this page (click on Show Advanced Settings to display all configuration items):

  • Default Caller ID: The Caller ID used for all the calls connected through this gateway, unless a more specific caller ID is specified for the user making the call.
  • Block Caller ID: If this is checked, all calls connected through this gateway will have caller ID blocked, unless a more specific caller ID is specified for the user.
  • Ignore user Caller ID: If this is checked, only the gateway Default Caller ID and Block Caller ID options are considered by this gateway.
  • Transform extension: If this is checked, the gateway will produce a caller ID by transforming the user extension using the rules for caller ID prefix and number of digits to keep. If not checked the caller ID specified for the user or for the gateway will be used.
  • Caller ID prefix: Optional prefix added to the user extension to create the caller ID.
  • Keep digits: The number of extension digits that are kept before adding the caller ID prefix. If the user extension has more digits than configured here, leading digits are dropped when creating the caller ID. The default value of 0 means keep all digits.
  • Specify Caller ID: If this is checked, the value of Caller ID below will overwrite any values specified for user and for gateway. Use this only if you need to specify the Display Name and optional URL parameters.
  • Caller ID: Used instead of the Default Caller ID specified above.
  • Display name: The name that appears on the phone when the call is received. Different gateways or ITSPs handle this field differently, so that actual results will vary. Some phones may not support displaying this value correctly.
  • URL parameters: Optional SIP URI parameters of the following form:key1=value1;key2= value2.

Click on the Apply button to keep any changes made on this page.

Dial Plan

The Dial Plan page, shown as follows, allows the administrator to add a dial prefix for outbound calls and also add the unmanaged gateway to any pre-configured system dialing rules.

Gateways in sipXecs 4.0: Part 2

The following configuration options are available on this page:

  • Prefix: The administrator can configure an outbound dialing prefix that will be added to all numbers for calls connected through this gateway. This is useful if centrex lines are in use.
  • Dialing Rules: The system dialing rules that should use this gateway can be specified in this section. Click on the More actions drop-down box to select the dial plan entries that are defined in the system. Click on the Apply button to keep any changes made on this page.

SIP Trunks

 

Increasingly, communications connectivity is being delivered to customers via SIP trunks across the Internet instead via traditional copper lines. A special gateway is included in the system that allows for connecting the sipXecs system to SIP trunks without any additional hardware required.

A SIP trunk can be added to the system in the Gateway screen by selecting the Add new gateway drop-down menu and cliking on SIP Trunk.

Gateways in sipXecs 4.0: Part 2

The following settings are available for SIP trunk gateways (click on Show Advanced Settings to see all of the SIP trunk's options):

  • Name: This is a descriptive name for the gateway.
  • Use provider template: Some pre-configured templates are available for popular SIP Trunk providers. Gateway settings will be pre-filled if your SIP trunking provider (ITSP) is on the list. To enter your own settings, or if your provider is not on the list, select "None". At the time of writing this, the available templates are: ATT, BT, Bandtel, bandwidth.com, CallWithUs, Cbeyond, Eutelia, LES.NET, sipcall.ch, Vitelity, VoIP User, and Voxitas.
  • Address: For a PSTN gateway—IP address of the gateway (example: 10.1.1.1), or the fully qualified hostname of the gateway (example: gateway. example.com). The gateway can be on any routed subnet without NAT. For a SIP trunking provider—External IP address or fully qualified hostname of the provider (for example, sip.provider.com).• Port: Optional UDP or TCP port if a non-standard port is used. Set to 0 to ignore this field (example: 5070).
  • Transport protocol: Set to UDP or TCP to force the SIP transport protocol. If set to "Auto", the transport is determined through a DNS query.
  • Location: The gateway use can be restricted by selecting a specific location for which it can be used. A location is represented by a group of users and you need to create a user group for every location that needs to be distinguished (remember that users can be in more than one group). This setting allows routing of calls based on in which location or from which user the call originates (source routing). This is useful if users located in a branch office would like to have a gateway preference so that calls are routed through their local gateway, for example, to preserve WAN bandwidth or to use caller ID offered by an analog gateway based on the PSTN number assigned to it. Only if that gateway is not available, will call routing fall back to other gateways specified for the corresponding dialing rule.
  • Shared: If this is checked, this gateway can be used by any user in any location, even if a specific location is selected. This setting is checked by default so that users in an identified location still use their preferred gateway, but the gateway can also be used by other users in other locations.
  • Description: This is used to document the gateway. Important information about the provider can be inserted here including tech support numbers or any other important and hard-to-locate information.
  • Route: How calls are routed to the SIP provider. Session Border Controller (SBC), SIP aware firewall or SIP proxy that processes calls directed at the provider served by this SIP tgrunk. Unless your system is on a public IP address, you will need an SBC. If in doubt, create an Internal SBC.

Summary

In this article, we have learnt all about gateways in sipXecs 4.0. We discussed how to add gateways which include managed gateways and unmanaged gateways. Managed gateways covered PSTN Lines, Caller ID, Dial Plan, SIP, Voice Codecs, Proxy and Registration, DTMF & Dialing, Advanced Parameters, Supplementary Services, FXO, Network, Media, RTP/RTPC and Management. Unmanaged gateways included Add gateway, Caller ID and Dial Plan. We also learnt about SIP Trunks towards the end of the discussion.

If you have read this article you may be interested to view :

About the Author :


Michael Peacock

Michael Peacock is a web developer from Newcastle, UK and has a degree in Software Engineering from the University of Durham. After meeting his business partner at Durham, he co-founded Peacock Carter, a Newcastle based creative consultancy specializing in web design, web development and corporate identity.

Michael loves working on web related projects. When he is not working on client projects, he is often tinkering with a web app of his own.

He has been involved with a number of books, having written two books himself (and working on his third): Selling online with Drupal e-Commerce Packt, and Building websites with TYPO3 Packt. He has also done technical reviews of two other books: Mobile Web Development Packt, and Drupal Education & E-Learning Packt.

You can follow Michael on Twitter.

Contact Michael Peacock

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