Gateways in sipXecs 4.0: Part 1

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

In this article by Michael W. Picher, we will learn all about gateways in sipXecs 4.0. We will discuss how to add gateways which will include managed gateways and unmanaged gateways. Managed gateways will include 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 will cover Add gateway, Caller ID and Dial Plan. We will also learn about SIP Trunks towards the end of the discussion.

Gateways provide the connectivity required to reach other systems. These systems can be other sipXecs PBX's, traditional phone lines, or Internet Telephony Service Providers (ITSPs).Connecting the IP phone system to the outside world is one of the most difficult tasks in making the phone system work. If the network infrastructure is configured properly for quality of service, the connection to the outside world can most likely be the source of any call quality problems.

Traditional analog Plain Old Telephone Service (POTS) lines are the largest source of frustration. If you can avoid them by utilizing a digital type of service or an ITSP, by all means take that avenue. For those not so lucky, you'll learn more about them then you ever thought you needed to. Typically, volume levels, line disconnect, and echo are the most common problems. Most gateways will have some advanced settings for dealing with these issues but they are different for every manufacturer.

Adding gateways

There are three types of gateways that can be configured to work in sipXecs; managed, unmanaged, and SIP Trunks. A managed gateway is a hardware device that connects to a traditional phone line. sipXecs knows how to generate configuration files (plug and play) for it. An unmanaged gateway is either a hardware device for which sipXecs doesn't know how to generate configuration files, or it may be another SIP PBX. A SIP Trunk is a connection to an ITSP.

Managed gateways

At present, there are eight gateways for which sipXecs generates configuration information (ACME 1000 and AudioCodes Models MP114, MP118, Mediant 1000/2000/3000/BRI, and TP260). This is just a small cross section of gateways available in the market. If your gateway is not in this list, see the following Unmanaged gateways subsection. The following detailed information about managed gateways may prove to be useful in setting up an unmanaged gateway.

For the following example screens, we'll utilize an AudioCodes MP114 FXO (Foreign Exchange Office) gateway. This particular gateway has four analog ports for connecting to POTS lines. Information on the gateway is available at http://www.audiocodes.com/products/mediapack-1xx. To  add the gateway, click on the Gateways menu item in the  Devices menu. As shown in the following screenshot, there are no gateways configured by default.

To add the gateway, click on the Gateways menu item in the Devices menu. As shown in the following screenshot, there are no gateways configured by default.

Gateways in sipXecs 4.0: Part 1

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

Gateways in sipXecs 4.0: Part 1

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

  • Name: A name 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 address 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.
  • Serial Number : This is the Ethernet MAC address of the gateway.
  • Firmware Version : Certain gateways may have different configuration file information or formats depending on the version of firmware in the device. Select the version of firmware that is loaded in the gateway (see manufacturer's documentation).
  • 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 the location or the user from which 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 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. Click on the gateway name to reveal more configuration options, as shown in the following screenshot:

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 gateway name to reveal more configuration options, as shown in the following screenshot:

Gateways in sipXecs 4.0: Part 1

In the following subsections, we'll explore the managed gateway settings available.

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PSTN Lines

Click on the PSTN Lines item on the left hand menu. As shown below, there are no lines defined on the gateway by default.

Gateways in sipXecs 4.0: Part 1

Click on the Add PSTN Line hyperlink and the following page will be displayed:

Gateways in sipXecs 4.0: Part 1

This page has two configuration options:

  • Automatic Dialing: If this is enabled, calls received on this PSTN line will be automatically sent to the destination (extension or user) specified below it. The default is enabled. If it is not enabled, incoming calls will ring and never be forwarded.
  • Extension: This is the destination extension for incoming calls on this PSTN line, such as an auto attendant, a hunt group, an ACD queue, or any internal extension, user, or alias. The default value is operator, which is an alias on the auto attendant.

Click on OK to add the line to the gateway. Repeat for the number of lines that are in use on the gateway.

Caller ID

The Caller ID page allows the administrator to set outbound caller ID information. This isn't particularly useful for analog lines as this is not a function that can be utilized. With a T1/PRI/BRI gateway, the administrator can insert whatever is necessary for caller ID information to the outgoing calls.

Gateways in sipXecs 4.0: Part 1

The following options are available on the preceding Caller ID page.

  • Default CallerID: This is the outgoing 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 outbound caller ID blocked, unless a more specific caller ID is specified for the user.
  • Ignore user CallerID: If this setting is checked, only the gateway Default Caller ID and Block Caller ID options are used for outgoing calls through this gateway.
  • Transform extension: If this setting is checked, the gateway will produce the caller ID by transforming the dialing user's extension using the rules for caller ID prefix and number of digits to keep. If it is not checked, the caller ID specified for the user or for the gateway will be used.
  • Caller ID prefix: This is an optional prefix added to the user extension to create the caller ID.
  • Keep digits: This is the number of user extension digits that are kept before adding the Caller ID prefix. If the user extension has more digits than configured, then the leading digits are dropped while creating the caller ID. The default value of 0 means keep all of the digits.

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

Dial Plan

The Dial Plan page, as shown in the following screenshot, allows the administrator to add a dial prefix for outbound calls and also add the gateway to any pre-configured system dialing rules (see the Dial Plan section under Unmanaged gateways, later in this article).

Gateways in sipXecs 4.0: Part 1

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

The SIP configuration page allows fine-grained control over SIP parameters in the gateway, if Show Advanced Settings is clicked. Most commonly, all of the default settings can be used.

Gateways in sipXecs 4.0: Part 1

The following configuration options are available on this page. Click on the Show Advanced Settings hyperlink to reveal all options. (For more detailed information about how all of these settings affect the gateway, please refer to the manufacturer's documentation.)

  • PRACK Mode: PRACK stands for Provisional Response Acknowledgement support for the gateway. PRACK is defined in RFC 3262. Provisional responses provide information on the progress of the request processing. The default setting for this mode is Supported (checked).
  • Channel Select Mode: This mode determines how the outbound phone lines are selected. The commonly used settings are Ascending (starting from line 1 and progressing up in line numbers, each time resetting to the lowest available number line that is not in use), Cyclic Ascending (starting from line 1 and ascending in line numbers for each call and wrapping around to line 1 again after each line has been used), Descending (starting at the highest line number available and descending to line 1, each time resetting to the highest available number line that is not in use) and Cyclic Descending (starting at the highest line number available and descending in line numbers for each call and wrapping around to line 1). The default value is Cyclic Ascending Note that it is best to use the opposite direction to show how the lines ring into the system from the outside provider's inbound hunt group. This helps to avoid a condition referred to as glare. Glare occurs if the phone system picks up a line to dial out that is ringing in at the same time.
  • Enable Early Media: Early Media refers to the ability of two SIP user agents to communicate before a SIP call is actually established, which is generally for call setup with a PSTN or PBX connection. The default is enabled (checked).
  • Asserted ID Mode: This mode defines the header that is used in the generated INVITE request. The header also depends on the calling privacy—it can be allowed or restricted. The default setting is disabled for this mode.
  • Tel URI for Asserted Identity: If "Asserted Identity" is selected, the telephone URI will be used for the AI. The default is to not use the Tel URI (unchecked).
  • Fax Signaling: Fax machines (and modems for that matter) have a difficult time communicating over a normal G.711 voice path. The T.38 protocol allows fax machines to communicate between gateways. The default is to have T.38 enabled.
  • Detect Fax on Answer Tone: If T.38 is enabled above, a fax machine can be detected when the call is answered with this setting. The default is T.38 on Preamble.
  • SIP Transport Type: This selects UDP, TCP, or TLS for the SIP transport type. The default is UDP.
  • UDP SIP Port: This allows the SIP UDP signaling port to be changed. The default is 5060.
  • TCP SIP Port: This allows the SIP TCP signaling port to be changed. The default is 5060.
  • TLS SIP Port: This allows the SIP TLS signaling port to be changed. The default is 5061.
  • TCP Connection Reuse: If TCP is reused, a SIP client behind a NAT'd network can keep a TCP connection open and both client and server can reuse it. This makes TCP communication potentially easier when NAT is in use. The default is to not reuse the TCP connection (unchecked).
  • Tel to IP No Answer Timeout: This is the timeout for a phone call ringing into a SIP extension. The default value is 180 seconds. This may need to be increased if the inbound calls need to ring longer.
  • Remote Party ID: This setting is not used by sipXecs but might be used in a gateway-to-gateway configuration. It allows the receiving gateway to identify a remote gateway by its ID. The default value is to not send it (unchecked).
  • RPI Header Content: Used with the above setting. The default is set to "Include Number Plan" and "Type".
  • History-Info Header: From RFS—a standard mechanism for capturing the history information associated with a Session Initiation Protocol (SIP) request. This capability enables many enhanced services by providing the information as to how and why a call arrives at a specific application or user. This information is not used by sipXecs and is disabled by default (unchecked).
  • Use Source Number as Display Name: This uses the source phone number as the display name on inbound calls to the PBX. The default setting is "No".
  • Use Display Name as Source Number: This uses the configured display name as the source number on inbound calls. The default is disabled (unchecked).
  • Play Ringback Tone to IP: Allows the gateway to play the ringing tone to the IP side of the gateway. The default is disabled (unchecked).
  • Play Ringback Tone to Tel: Allows the gateway to pick up a phone call and then continue playing a ring tone to the caller. Valid settings are "Don't Play", "Always Play", "Play" according to 180/183, and "Play" according to PI (default). See Audiocodes manual for more detailed information.
  • Enable GRUU: A URI that routes to a specific UA instance is called a Globally Routable UA URI(GRUU). The default is disabled (unchecked). See Audiocodes manual for more detailed information.
  • User-Agent Information : Defines the string that is used in the SIP request header "User-Agent" and SIP response header "Server". If it is not configured, the default string "AudioCodes product name s/w-version" is used. The default is blank (not configured).
  • SDP Session Owner: The value of the Session Owner line ("o" field) in outgoing Session Description Protocol (SDP) bodies. May be up to 39 characters. The default value is AudiocodesGW.
  • Subject: Defines the value of the subject header in outgoing INVITE messages. If it is not specified, the subject header isn't included. The default is blank (not specified).
  • Multiple Packetization Time Format: This setting enables the IP gateway to define a separate Packetization period for each negotiated coder in the SDP. The "mptime" attribute is included only if this parameter is enabled, even if the remote side includes it in the SDP offer. The default is disabled (unchecked).
  • Reason Header: This enables or disables the use of the SIP reason header. The default is enabled (checked).
  • 3xxBehavior: Determines the gateway's behavior when a SIP 3xx response is received for an outgoing INVITE request. The gateway can either use the same call identifiers (Call ID, branch, to and from tags) or change them in the new initiated INVITE. If this is disabled, the gateway will use different call identifiers for a redirected INVITE message. If it is enabled, the gateway will use the same call identifiers. The default is set to disabled (unchecked).
  • Enable P-Charging-Vector: From RFC 3455—P-Headers are a set of private Session Initiation Protocol (SIP) headers (P-headers) used by the 3rd-Generation Partnership Project (3GPP). The P-Charging-Vector header is used to convey charging related information, such as the globally unique IMS Charging Identifier (ICID) value. This value is used primarily for call accounting but not used by sipXecs and thus the default is set to disabled (unchecked).
  • Enable Voicemail URI: This enables or disables the interworking of target and cause for redirection from Tel to IP and vice versa, according to RFC 4468. The default is disabled (unchecked).
  • SIP T1 Retransmission Timer [msec]: This is the time interval (in milliseconds) between the first transmission of a SIP message and the first retransmission of the same message. The default is 500 ms.
  • SIP T2 Retransmission Timer [msec]: This is the maximum interval (in milliseconds) between retransmissions of SIP messages. The time interval between subsequent retransmissions of the same SIP message starts with SIP T1 Retransmission Timer and is multiplied by two until the SIP T2 Retransmission Timer is reached. The default is 4000.
  • SIP Maximum RTX: This is the number of UDP transmissions (first transmission + retransmissions) of SIP message. The range is 1 to 7. The default value is 7.

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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Voice Codecs

The Voice Codecs page allows the administrator to specify what Codecs (compression / decompression algorithms) the gateway is allowed to use. By default, the sipXecs media services only support G.711 uLaw and a Law Codecs (64 to 80 Kbps per call).

Gateways in sipXecs 4.0: Part 1

From the drop-down boxes select the protocols required on the gateway. If a remote phone is being used for the outgoing calls through this gateway, it may be advantageous to select G.729 as the second Codec (8 to 12 Kbps per call). Click on the Apply button to keep any changes made on this page.

Proxy and Registration

The Proxy and Registration page determines how the gateway interacts with the PBX. In the sipXecs world, gateways do not register with the PBX (however, individual lines on an FXS gateway do).

Gateways in sipXecs 4.0: Part 1

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

  • Proxy IP Address: This should be set to the SIP domain of the PBX. (With the example system that has been used, this would be xyzcompany.com.)
  • Proxy Keepalive Mode: If this is set to "Options", a SIP OPTIONS message is sent every Proxy Keep Alive Time. If set to "Register", a SIP REGISTER message is sent every registration time. Any response from the proxy, either success (200 OK) or failure (4xx response) is considered as if the proxy is correctly communicating. Since gateways don't need to register with sipXecs, this value is not needed and is set to disabled by default.
  • Proxy Keep Alive Time: This defines the proxy keep-alive time interval (in seconds) between keep-alive messages. The default is set to 5 seconds but is not used because Proxy Keepalive Mode is set to disabled.
  • Send All INVITE to Proxy: If this is disabled, INVITE messages resulting from redirect or transfer will be sent directly to the destination URI. The default is set to enabled (checked) so that all INVITE messages are transmitted to the proxy server (to the PBX).
  • Gateway Name: This is the name of the gateway. Ensure that the name you choose is the one that the Proxy is configured with to identify your media gateway. The default is set to the SIP domain name (from the example that has been used it would be set to xyzcompany.com).
  • DNS Query Type: This is for querying the SIP domain. It can be set to A-Record, DNS NAPTR, or SRV records. The default setting is set to SRV.
  • Gateway Name for OPTIONS: The OPTIONS Request-URI host part contains either the gateway's IP address or a string defined by the parameter "Gateway Name". If this is enabled, the gateway sends its name as the host part of the SIP URI. If it is disabled, the IP address is used. The default is enabled (checked).
  • Hot-Swap Redundancy Mode: The user can enable or disable proxy hot-swap redundancy mode. This allows SIP INVITE/REGISTER messages to be routed to a redundant proxy/registrar server if the primary proxy/registrar server does not respond. This is not required by sipXecs (unchecked) because SRV records are used for routing between redundant SIP proxies.
  • Number of RTX before Hot-Swap: The AudioCodes default value is 3 retransmits before switching to a redundant SIP Proxy. This setting is not needed either because of the use of SRV records.
  • Challenge Caching Mode: To reduce the number of SIP messages, challenges can be cached. Select the desired level of challenge caching to use with SIP proxies. The default is set to none (challenges are not cached).
  • Mutual Authentication Mode: This mode selects whether Authentication and Key Agreement (AKA) digest authentication information are optional or mandatory for incoming requests. If it is set to "Mandatory", incoming requests without AKA information will be rejected. The default setting is "Optional".

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

DTMF & Dialing

The Gateway DTMF & Dialing page is used to configure how the gateway interacts with dialed digits.

Gateways in sipXecs 4.0: Part 1

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

  • Dialed Digits Max. Length: The maximum dialed number length allowed. The default is set to 14 digits.
  • Inter-Digit Timeout: The gateway's dial time-out before end of dialed string is automatically set. The default is 4 seconds.
  • Declare RFC2833 in SDP: RFC2833 determines how dialed digits (DTMF) are dealt with within a phone call. The gateway needs to be set to declare RFC2833 in the Session Description Protocol (SDP  ). DTMF is done in-band with the media services.
  • TxDTMF Option: Transmit DTMF options in order of priority (all are set to no negotiation).
  • RFC2833 DTMF Payload Type: The RFC2833 DTMF Relay dynamic payload type. Allowed ranges are: 96-99 and 106-27. The default value is 96.
  • Flash Hook Detection: Reported Period—the flash-hook period (in milliseconds) that is reported to the FXO port. The default value is 790 ms.

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

>> Continue Reading Gateways in sipXecs 4.0: Part 2

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About the Author :


Michael W. Picher

As an industry veteran with over 20 years in Information Technology consulting, Michael brings a network engineer’s perspective to the Telephony business. After receiving a Bachelor of Science degree in Computer Engineering from the University of Maine, Michael worked hard to build up a computer manufacturing business which he left in the mid ‘90s. Following the manufacturing endeavor Michael worked with two close friends to build what became one of Maine’s largest home-grown technology consulting and software development firms. After successfully selling the consulting business to a large out of state firm, Michael turned his attention to the growing IP Telephony space. Michael has helped successfully deploy some of the region’s largest IP based communications systems and the infrastructure required to support those systems.

Away from technology, Michael enjoys life with his wife Debra and son Matthew on their large Maine wild blueberry farm in rural Maine. Snowmobiling and hunting are the family choices for fun and Michael is also a long time Autocross fanatic with multiple class wins in his beloved Mini Cooper S.

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