Audiocodes MP-11X
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AUDIOCODES Acts 24x7 Support MP11X S2 yea #ACTS24X7MP11X_S2YRDetails
Brand: AUDIOCODES
Part Numbers: ACTS24X7-MP11XS2YR, ACTS24X7-MP11X_S2/YR, ACTS24X7MP11X_S2YR
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Manual
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(French)Audiocodes MP-11X - Guide D Installation Rapide, size: 573 KB |
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(English)Check if your language version is avaliable. Most of manuals are avaliable in many languages. |
Audiocodes MP-11X
Video review
AudioCodes MediaPack MP 11X Product Demo
User reviews and opinions
| reggoboy |
11:24pm on Thursday, October 14th, 2010 ![]() |
| A universal adapter for the USB cable would be a great addition. Proprietary connections are a pain. Great capacity, great display. Sadly I Scratched the screen already. But I ordered a hard plastic cover for it which I highly recommend if you plan on using it a lot. | |
| Larsen |
10:00am on Tuesday, October 12th, 2010 ![]() |
| I have had a Sony WMA MP3 player for a year or so, and that is my baseline. On the Sansa Fuze - the biggest puzzle is the battery level indicator. Bought this for my daughter. She needed something simple. Turns out it has a great user-interface with a fabulous sound. | |
| schmolch |
9:55pm on Thursday, September 2nd, 2010 ![]() |
| Great MP3 Player Who needs an iPod when you can get this? Not bad but voice recording function is not as good as I expected, unexpected digital circuit noise involved. Great easy to use product! When I was shopping for an MP3 player, I wanted one that was reliable and very easy to use. | |
| aldaba |
5:38pm on Saturday, August 28th, 2010 ![]() |
| I use the Fuze at the beach and when I am reading. It has a great sound and is very,very easy to navigate. Exactly as described - an awesome player. Had it 6 months now and not a single problem. Very easy to setup and expandable too. I enjoy it a lot. I really like the shape and its really good quality. The sound is crisp and clear. "Good shape","Durable","High Quality","Good Value" "Not ipod" | |
| Proporciya |
4:33am on Tuesday, August 3rd, 2010 ![]() |
| manual said 5 hrs.of play,more like 2.5 hrs.barely makes the journey to work.headphones are a joke. Great MP3 player. Easy To Set Up","Long Battery Life","Lots of Storage","Reliable Performance Confusing Controls","Poor Sound Quality | |
| xgecko |
11:21pm on Monday, July 5th, 2010 ![]() |
| Does everything I want and more. I can delete files without hooking to a computer and increase capacity with a mini sd card. This is product is great. Nice head phones for stock. easy to use and Super cheap. everyone likes cheap and upgradable This product was magnifacent in all areas its a good quality and works really well, would buy it again and recomend to others! | |
| eBusiness |
8:30pm on Thursday, March 25th, 2010 ![]() |
| Received product in timely manner. Product has performed as expected. 24 hours on a charge took it camping and lasted all weekend on a single charge I wouldt use it for video or pictures as you need to convert to view th... | |
Comments posted on www.ps2netdrivers.net are solely the views and opinions of the people posting them and do not necessarily reflect the views or opinions of us.
Documents

MediaPack SIP URL: sip:phone number@IP address (such as 1225556@10.1.2.4, where 122556 is the phone number of the source or destination) or sip:phone_number@domain name, such as 122556@myproxy.com. Note that the SIP URI host name can be configured differently per called number. Supports RFC 4040, RTP payload format for a 64 kbit/s transparent data. Can negotiate coder from a list of given coders. Supports negotiation of dynamic payload types. Supports multiple ptime values per coder. Supports RFC 3389, RTP Payload for Comfort Noise. Supports RFC 3824, Using E.164 numbers with SIP (ENUM). Supports reception and DNS resolution of FQDNs received in SDP. Supports <draft-ietf-sip-gruu-09>, Obtaining and Using Globally Routable User Agent (UA) URIs (GRUU) in SIP Responds to OPTIONS messages both outside a SIP dialog and in mid-call. Generates SIP OPTIONS messages as Proxy keep-alive mechanism. Publishes the total number of free Tel channels in a 200 OK response to an OPTIONS requests. Implementation of MWI IETF <draft-ietf-sipping-mwi-04.txt>, including SUBSCRIBE (to the MWI server). The MediaPack FXS gateways can accept an MWI NOTIFY message that indicates waiting messages or indicates that the MWI is cleared. Supports 3-Way Conference using an external media server. For more updated information on the gateways supported features, refer to the latest MP-11x & MP-124 SIP Release Notes.
2. MediaPack Physical Description
MediaPack Physical Description
This section provides detailed information on the hardware, the location and functionality of the LEDs, buttons and connectors on the front and rear panels of the MP-11x (refer to Section 2.1 below) and MP-124 (refer to Section 2.2 on page 27) gateways. For detailed information on installing the MediaPack, refer to Chapter 0 on page 29.
MP-11x Physical Description
MP-11x Front Panel
Figure 2-1 illustrates the front layout of the MP-118 (almost identical on MP-114 and MP112). Table 2-1 lists and describes the front panel LEDs on the MP-11x.
MP-11x (FXS/FXO) gateways feature similar front panel LEDs; they only differ in the number of Channels Status LEDs, which correspond to the number of channels. Figure 2-1: MP-118 Front Panel Connectors
MediaPack Table 2-1: Definition of MP-11x Front Panel LED Indicators (continues on pages 26 to 26) LED
Channels Status
Telephone Interface
Blinking Fast Blinking Off On
Definition
The phone is ringing (incoming call, before answering). Line malfunction Normal onhook position Offhook Ringing Valid 10/100 Base-TX Ethernet connection No uplink Failure (fatal error). Or system initialization. Normal working condition Device powered, self-test OK Software loading or System failure Power iscurrently being supplied to the device Either theres a failure / disruption in the AC power supply or power is currently not being supplied to the device through the AC power supply entry.
Table 3-1: View of the MP-11x Base Item # 3 Functionality Square slot used to attach anti-slide bumpers (for desktop mounting) Oval notch used to attach the MP-11x to a wall Screw opening used to attach the MP-11x to a 19-inch shelf rack
3.1.4.1
Mounting the MP-11x on a Desktop
Attach the four (supplied) anti-slide bumpers to the base of the MP-11x (refer to item #1 in Figure 3-2) and place it on the desktop in the position you require.
3.1.4.2
Mounting the MP-11x on a Wall To mount the MP-11x on a wall, take these 4 steps:
1. Drill four holes according to the following dimensions: 2. 3. 4. Side-to-side distance 140 mm. Front-to-back distance 101.4 mm. Insert a wall anchor of the appropriate size into each hole. Fasten a DIN 96 3.5X20 wood screw (not supplied) into each of the wall anchors. Position the four oval notches located on the base of the MP-11x (refer to item #2 in Figure 3-2) over the four screws and hang the MP-11x on them.
3.1.4.3
Installing the MP-11x in a 19-inch Rack
The MP-11x can be installed in a standard 19-inch rack by placing it on an AudioCodes' 19-inch rack-mounting shelf that is pre-installed in the rack. The shelf can hold up to two MP-11x gateways. This shelf can be ordered separately from AudioCodes. The 19-inch rack shelf is not supplied in the standard package kit, but can be ordered separately (Bulk Pack package MCMK00015 with 10 rack mounting shelves for MP-11x). For ordering and pricing, please contact your AudioCodes' distributor.
Figure 3-3: MP-11x Rack Mount
Table 3-2: MP-11x Rack Mount Item # Functionality Standard rack holes used to attach the shelf to the rack Eight shelf-to-device screws
To install the MP-11x in a 19-inch rack, take these 3 steps:
1. 2. 3. Use the shelf-to-device screws (supplied) to attach one or two MP-11x devices to the shelf. Position the shelf in the rack and line up its side holes with the rack frame holes. Use four standard rack screws (not supplied) to attach the shelf to the rack.
Lifeline Tip Tip Ring Lifeline Ring
To cable the MP-11x/FXS Lifeline, take these 3 steps:
1. 2. 3. Connect the Lifeline splitter to port #1 on the MP-11x (the Lifeline splitter is a special order option). Connect the Lifeline phone to Port A on the Lifeline splitter. Connect an analog PSTN line to Port B on the Lifeline splitter.
To cable the combined MP-11x FXS/FXO Lifeline, take these 2 steps:
1. 2. Connect a fax machine, modem, or phone to each of the FXS ports. Connect an analog PSTN line to each of the FXO ports.
The use of the Lifeline on network failure can be disabled using the LifeLineType ini file parameter (described in Table 5-55 on page 182).
Installing the MP-124
To install the MP-124, take these 4 steps:
1. Unpack the MP-124 (refer to Section 3.2.1 below). 2. Check the package contents (refer to Section 3.2.2 below). 3. Mount the MP-124 (refer to Section 3.2.3 on page 36). 4. Cable the MP-124 (refer to Section 3.2.4 on page 38). After connecting the MP-124 to the power source, the Ready and LAN LEDs on the front panel turn to green (after a self-testing period of about 1 minute). Any malfunction changes the Ready LED to red. When you have completed the above relevant sections you are then ready to start configuring the gateway (Chapter 4 on page 41).
To unpack the MP-124, take these 6 steps:
1. 2. 3. 4. 5. 6. Open the carton and remove packing materials. Remove the MP-124 gateway from the carton. Check that there is no equipment damage. Check, retain and process any documents. Notify AudioCodes or your local supplier of any damage or discrepancies. Retain any diskettes or CDs.
Ensure that in addition to the MP-124, the package contains: AC power cable. 2 short equal-length brackets and bracket-to-device screws for the 19-inch rack installation. A CD with software and documentation may be included. The MediaPack Fast Track Installation Guide.
Mounting the MP-124
The MP-124 can be mounted on a desktop or installed in a standard 19-inch rack. Refer to Section 3.2.4 on page 38 for cabling the MP-124.
3.2.3.1
Mounting the MP-124 on a Desktop
No brackets are required. Simply place the MP-124 on the desktop in the position you require. Figure 3-9: Desktop or Shelf Mounting
Rack Mount Safety Instructions (UL)
When installing the chassis in a rack, be sure to implement the following Safety instructions recommended by Underwriters Laboratories: Elevated Operating Ambient - If installed in a closed or multi-unit rack assembly, the operating ambient temperature of the rack environment may be greater than room ambient. Therefore, consideration should be given to installing the equipment in an environment compatible with the maximum ambient temperature (Tma) specified by the manufacturer. Reduced Air Flow - Installation of the equipment in a rack should be such that the amount of air flow required for safe operation on the equipment is not compromised. Mechanical Loading - Mounting of the equipment in the rack should be such that a hazardous condition is not achieved due to uneven mechanical loading. Circuit Overloading - Consideration should be given to the connection of the equipment to the supply circuit and the effect that overloading of the circuits might have on overcurrent protection and supply wiring. Appropriate consideration of equipment nameplate ratings should be used when addressing this concern. Reliable Earthing - Reliable earthing of rack-mounted equipment should be maintained. Particular attention should be given to supply connections other than direct connections to the branch circuit (e.g., use of power strips.)
Note: Digit Mapping Rules can be used instead.
Inter Digits Timeout for Overlap Dialing [sec] [TimeBetweenDigits]
Table 5-7: DTMF & Dialing Parameters (continues on pages 74 to 76)
Declare RFC 2833 in SDP [RxDTMFOption]
Defines the supported Receive DTMF negotiation method. No [0] = Dont declare RFC 2833 telephony-event parameter in SDP Yes [3] = Declare RFC 2833 telephony-event parameter in SDP (default) The MediaPack is designed to always be receptive to RFC 2833 DTMF relay packets. Therefore, it is always correct to include the telephony-event parameter as a default in the SDP. However some gateways use the absence of the telephony-event from the SDP to decide to send DTMF digits in-band using G.711 coder, if this is the case you can set RxDTMFOption=0.
1st to 5th Tx DTMF Option [TxDTMFOption]
Determines a single or several preferred transmit DTMF negotiation methods. 0 (Not Supported) = No negotiation, DTMF digits are sent according to the parameters DTMFTransportType and RFC2833PayloadType (default). 1 (INFO Nortel) = Sends DTMF digits according to IETF <draft-choudhuri-sip-infodigit-00>. 2 (NOTIFY) = Sends DTMF digits according to <draft-mahy-sipping-signaleddigits-01>. 3 (INFO Cisco) = Sends DTMF digits according to Cisco format. 4 (RFC 2833).
Note 1: DTMF negotiation methods are prioritized according to the order of their appearance. Note 2: When out-of-band DTMF transfer is used (1, 2 or 3), the parameter DTMFTransportType is automatically set to 0 (DTMF digits are erased from the RTP stream). Note 3: When RFC 2833 (4) is selected, the gateway: Negotiates RFC 2833 Payload Type (PT) using local and remote SDPs. Sends DTMF packets using RFC 2833 PT according to the PT in the received SDP. Expects to receive RFC 2833 packets with the same PT as configured by the parameter RFC2833PayloadType. Uses the same PT for send and receive if the remote party doesnt include the RFC 2833 DTMF PT in its SDP. Note 4: When TxDTMFOption is set to 0, the RFC 2833 PT is set according to the parameter RFC2833PayloadType for both transmit and receive. ini file note: The DTMF transmit methods are defined using a repetition of the same (TxDTMFOption) parameter (up to five options can be provided).
RFC 2833 Payload Type [RFC2833PayloadType] The RFC 2833 DTMF relay dynamic payload type. Range: 96 to 99, 106 to 127; Default = 96 The 100, 102 to 105 range is allocated for proprietary usage. Note 1: Cisco is using payload type 101 for RFC 2833. Note 2: When RFC 2833 payload type (PT) negotiation is used (TxDTMFOption=4), this payload type is used for the received DTMF packets. If negotiation isnt used, this payload type is used for receive and for transmit. Supported hook-flash Transport Type (method by which hook-flash is sent and received). Valid options include: Not Supported [0] = Hook-Flash indication isnt sent (default). INFO [1] = Send proprietary INFO message with Hook-Flash indication. RFC 2833 [4] = RFC 2833. Note: FXO gateways support the receiving of RFC 2833 Hook-Flash signals. This parameter is obsolete; use instead the parameter HookFlashOption
Enable Hold [EnableHold]
No [0] = Disable the Hold service. Yes [1] = Enable the Hold service (default). If the Hold service is enabled, a user can activate Hold (or Unhold) using the hookflash. On receiving a Hold request, the remote party is put on-hold and hears the hold tone. Note: To use this service, the gateways at both ends must support this option. Determines the format of the hold request. [0] = The connection IP address in SDP is 0.0.0.0 (default). 0.0.0.0 Send Only [1] = The last attribute of the SDP contains the following a=sendonly. No [0] = Disable the call transfer service. Yes [1] = Enable the call transfer service (using REFER) (default). If the Transfer service is enabled, the user can activate Transfer using hook-flash signaling. If this service is enabled, the remote party performs the call transfer. Note 1: To use this service, the gateways at both ends must support this option. Note 2: To use this service, set the parameter Enable Hold to Yes. Defined string that is added, as a prefix, to the transferred / forwarded called number, when REFER / 3xx message is received. Note 1: The number manipulation rules apply to the user part of the REFER-TO / Contact URI before it is sent in the INVITE message. Note 2: The xferprefix parameter can be used to apply different manipulation rules to differentiate the transferred / forwarded number from the original dialed number. No [0] = Disable the Call Forward service. Yes [1] = Enable Call Forward service (using REFER) (default). For FXS gateways a Call Forward table must be defined to use the Call Forward service. To define the Call Forward table, refer to Section 5.5.9.4 on page 122. Note: To use this service, the gateways at both ends must support this option. No [0] = Disable the Call Waiting service. Yes [1] = Enable the Call Waiting service (default). If enabled, when an FXS gateway receives a call on a busy endpoint, it responds with a 182 response (and not with a 486 busy). The gateway plays a call waiting indication signal. When hook-flash is detected, the gateway switches to the waiting call. The gateway that initiated the waiting call plays a Call Waiting Ringback tone to the calling party after a 182 response is received. Note 1: The gateways Call Progress Tones file must include a call waiting Ringback tone (caller side) and a call waiting tone (called side, FXS only). Note 2: The Enable Hold parameter must be enabled on both the calling and the called sides. For information on the Call Waiting feature, refer to Section 8.1.5 on page 217. For information on the Call Progress Tones file, refer to Section 15.1 on page 325.
Hold Format [HoldFormat]
Enable Transfer [EnableTransfer]
Destination Phone Prefix
Source Phone Prefix
Any telephone number whose destination number matches the prefix defined in the Destination Phone Prefix field and its source number matches the prefix defined in the adjacent Source Phone Prefix field, is sent to the IP address entered in the IP Address field. Note that Tel to IP routing can be performed according to a combination of source and destination phone prefixes, or using each independently.
Note 1: An additional entry of the same prefixes can be assigned to enable alternative routing. Note 2: For available notations that represent multiple numbers, refer to Section 5.5.3.1 on page 95.
Destination IP Address
In each of the IP Address fields, enter the IP address (and optionally port number) that is assigned to these prefixes. Domain names, such as domain.com, can be used instead of IP addresses. For example: <IP Address>:<Port> To discard outgoing IP calls, enter 0.0.0.0 in this field. Note: When using domain names, you must enter a DNS server IP address, or alternatively define these names in the Internal DNS Table. Enter the number of the IP profile that is assigned to the destination IP address defined in the Destination IP Address field. A read only field representing the quality of service of the destination IP address. N/A = Alternative Routing feature is disabled. OK = IP route is available Ping Error = No ping to IP destination, route is not available QoS Low = Bad QoS of IP destination, route is not available DNS Error = No DNS resolution (only when domain name is used instead of an IP address). An optional Charge Code (1 to 25) can be applied to each routing rule to associate it with an entry in the Charge Code table (refer to Section 5.5.2.3.1 on page 88). Prefix = <Destination Phone Prefix>, <IP Address>,<Src Phone Prefix>,<IP Profile ID>,<Charge Code> For example: Prefix = 20,10.2.10.2,202,1,15 Prefix = 10[340-451]xxx#,10.2.10.6,*,1,1 Prefix = *,gateway.domain.com,*,20 Note 1: <destination / source phone prefix> can be single number or a range of numbers. For available notations, refer to Section 5.5.3.1 on page 95. Note 2: This parameter can appear up to 50 times. Note 3: Parameters can be skipped by using the sign $$, for example: Prefix = $$,10.2.10.2,202,1 Note 4: An optional IP ProfileID (1 to 9) can be applied to each routing rule.
DNS2IP
DNS2IP = <Hostname>, <first IP address>, <second IP address> For example: DNS2IP = Domainname.com, 10.8.21.4, 10.13.2.95
Note: This parameter can appear up to 10 times.
5.5.5.5
Internal SRV Table
The Internal SRV table is used for resolving host names to DNS A-Records. Three different A-Records can be assigned to a hostname. Each A-Record contains the host name, priority, weight, and port.
If the Internal SRV table is configured, the gateway first tries to resolve a domain name using this table. If the domain name isnt found, the gateway performs an SRV resolution using an external DNS server.
To configure the Internal SRV table, take these 9 steps:
1. Open the Internal SRV Table screen (Protocol Management menu > Routing Tables submenu > Internal SRV Table option); the Internal SRV Table screen is displayed. Figure 5-20: Internal SRV Table Screen
2. 3. 4. 5. 6. 7. 8. 9.
In the Domain Name field, enter the hostname to be translated. You can enter a string up to 31 characters long. From the 'Transport Type' drop-down list, select a transport type. In the DNS Name 1' field, enter the first DNS A-Record to which the hostname is translated. In the Priority, Weight and Port fields, enter the relevant values\ Repeat steps 4 to 5, for the second and third DNS names, if required. Repeat steps 2 to 6, for each Internal SRV Table entry. Click the Submit button to save your changes. To save the changes so they are available after a hardware reset or power fail, refer to Section 5.10.2 on page 205. Table 5-20: Internal SRV ini File Parameter
SRV2IP
SRV2IP = <Internal Domain Name>, <Transport Type>, <DNS Name 1>, <Priority 1>, <Weight 1>, <Port 1>, <DNS Name 2>, <Priority 2>, <Weight 2>, <Port 2>, <DNS Name 3>, <Priority 3>, <Weight 3>, <Port 3> Note 1: If the internal SRV table is configured, the gateway first tries to resolve a domain name using this table. If the domain name isnt found, the gateway performs an SRV resolution using an external DNS server. Note 2: This parameter can appear up to 10 times.
5.5.5.6
Reasons for Alternative Routing
The Reasons for Alternative Routing screen includes two tables (Tel IP and IP Tel). Each table enables you to define up to 4 different release reasons. If a call is released as a result of one of these reasons, the gateway tries to find an alternative route to that call. The release reason for IP Tel calls is provided in Q.931 notation. The release reason for Tel IP calls is provided in SIP 4xx, 5xx and 6xx response codes. For Tel IP calls an alternative IP address, for IP Tel calls an alternative hunt group. Refer to 5.5.5.2 on page 100 for information on defining an alternative IP address. Refer to the 5.5.5.3 on page 102 for information on defining an alternative hunt group. You can use this table for example: For Tel IP calls, when there is no response to an INVITE message (after INVITE retransmissions), and the gateway then issues an internal 408 No Response implicit release reason. For IP Tel calls, when the destination is busy, and release reason #17 is issued or for other call releases that issue the default release reason (#3). Refer to DefaultReleaseCause in Table 5-8.
Activity Types to Report via 'Activity Log' Messages The Activity Log mechanism enables the MediaPack to send log messages (to a Syslog server) that report certain types of Web actions according to a pre-defined filter. The following filters are available:
Parameters Value Change [ActivityListToLog = PVC] Auxiliary Files Loading [ActivityListToLog = AFL] Device Reset [ActivityListToLog = DR] Flash Memory Burning [ActivityListToLog = FB] Device Software Update [ActivityListToLog = SWU] Changes made on-the-fly to parameters. Loading of auxiliary files (e.g., via Certificate screen). Device reset via the 'Maintenance Actions' screen. Burning of files / parameters to flash (e.g., 'Maintenance Actions' screen). cmp loading via the Software Upgrade Wizard.
Access to Restricted Domains [ActivityListToLog = ARD]
Access to Restricted Domains. The following screens are restricted: (1) ini parameters (AdminPage) (2) General Security Settings (3) Configuration File (4) IPSec/IKE tables (5) Software Upgrade Key (6) Internal Firewall (7) Web Access List. (8) Web User Accounts Attempt to access the Embedded Web Server with a false / empty username or password. Changes made to sensitive parameters: (1) IP Address (2) Subnet Mask (3) Default Gateway IP Address (4) ActivityListToLog
Non Authorized Access [ActivityListToLog = NAA] Sensitive Parameters Value Change [ActivityListToLog = SPC]
ini file example: ActivityListToLog = 'pvc', 'afl', 'dr', 'fb', 'swu', 'ard', 'naa', 'spc'
5.6.6.1
Configuring the SNMP Managers Table
The SNMP Managers table allows you to configure the attributes of up to five SNMP managers.
To configure the SNMP Managers Table, take these 5 steps:
1. 2. Access the Management Settings screen (Advanced Configuration menu > Management Settings); the Management Settings screen is displayed (Figure 5-51). Open the SNMP Managers Table screen by clicking the arrow sign (-->) to the right of the SNMP Managers Table label; the SNMP Managers Table screen is displayed (Figure 5-52). Configure the SNMP Managers parameters according to Table 5-52. Click the Submit button to save your changes. To save the changes so they are available after a power fail, refer to Section 5.10.2 on page 205. Figure 5-52: SNMP Managers Table Screen
Not Connected (FXO only) Indicates that no analog line is connected to this port. Handset Offhook Indicates this channel is offhook but there is no active RTP session.
To monitor the details of a specific channel, take these 3 steps:
1. 2. 3. Click the numbered port icon of the specific channel whose detailed status you need to check/monitor; a shortcut menu appears. From the shortcut menu, choose Port Settings; the channel-specific Channel Status screen appears, shown in Figure 5-62. Click the submenu links to check/view a specific channels parameter settings. Figure 5-62: Channel Status Details Screen
Adding a Port Description
The 'Channel Status' screen allows you to add a brief text description or name for each port / channel.
To add a port description, take these 3 steps:
1. 2. 7. Open the Channel Status screen by clicking the Home icon. Click a port / channel icon, and then from the shortcut menu, choose Update Port Info; a text box appears. In the text box, type a brief description of this port, and then click Apply Port Info.
Resetting a Channel
The 'Channel Status' screen allows you to inactivate (reset) a channel. This is sometimes useful in cases, for example, when the gateway (FXO) is connected to a PBX and the communication between the two can't be disconnected (e.g., using reverse polarity).
To reset channel, take these 2 steps:
1. 2. Open the Channel Status screen by clicking the Home icon. Click a channel icon, and then from the shortcut menu, choose Reset Channel; the channel is changed to inactive. 196 Document #: LTRT-65408
Software Update
The Software Update menu enables users to upgrade the MediaPack software by loading a new cmp file along with the ini and a suite of auxiliary files, or to update the existing auxiliary files. The Software Update menu comprises two submenus: Software Upgrade Wizard (refer to Section 5.9.1 below). Load Auxiliary Files (refer to Section 5.9.2 on page 202).
When upgrading the MediaPack software you must load the new cmp file with all other related configuration files.
Software Upgrade Wizard
The Software Upgrade Wizard guides users through the process of software upgrade: selecting files and loading them to the gateway. The wizard also enables users to upgrade software while maintaining the existing configuration. Using the wizard obligates users to load and burn a cmp file. Users can choose to also use the Wizard to load the ini and auxiliary files (e.g., Call Progress Tones) but this option cannot be pursued without loading the cmp file. For the ini and each auxiliary file type, users can choose to reload an existing file, load a new file or not load a file at all. The Software Upgrade Wizard requires the MediaPack to be reset at the end of the process, disrupting any of its traffic. To avoid disruption, disable all traffic on the MediaPack before initiating the Wizard. Verify, prior to clicking the Start Software Upgrade button that no traffic is running on the device. After clicking this button a device reset is mandatory. Even if you choose to cancel the process in the middle, the device resets itself and the previous configuration burned to flash is reloaded.
Using DHCP
When the gateway is configured to use DHCP (DHCPEnable = 1), it attempts to contact the local DHCP server to obtain the networking parameters (IP address, subnet mask, default gateway, primary/secondary DNS server and two SIP server addresses). These network parameters have a time limit. After the time limit expires, the gateway must renew its lease from the DHCP server. Note that if the DHCP server denies the use of the gateway's current IP address and specifies a different IP address (according to RFC 1541), the gateway must change its networking parameters. If this happens while calls are in progress, they are not automatically rerouted to the new network address (since this function is beyond the scope of a VoIP gateway). Therefore, administrators are advised to configure DHCP servers to allow renewal of IP addresses.
If the gateway's network cable is disconnected and reconnected, a DHCP renewal is performed (to verify that the gateway is still connected to the same network).
When DHCP is enabled, the gateway also includes its product name (e.g., MP-118 FXS) in the DHCP option 60 Vendor Class Identifier. The DHCP server can use this product name to assign an IP address accordingly. After power-up, the gateway performs two distinct DHCP sequences. Only in the second sequence, DHCP option 60 is contained. If the gateway is reset from the Web/SNMP, only a single DHCP sequence containing option 60 is sent.
If DHCP procedure is used, the new gateway IP address, allocated by the DHCP server, must be detected.
If, during operation, the IP address of the gateway is changed as a result of a DHCP renewal, the gateway is automatically reset.
To detect the gateways IP address, follow one of the procedures below:
Starting with Boot version 1.92, the gateway can use a host name in the DHCP request. The host name is set to acl_nnnnn, where nnnnn stands for the gateways serial number (the serial number is equal to the last 6 digits of the MAC address converted from Hex to decimal). If the DHCP server registers this host name to a DNS server, the user can access the gateway (through a Web browser) using a URL of http://acl_<serial number> (instead of using the gateways IP address). For example, if the gateways MAC address is 00908f010280, the DNS name is acl_66176. After physically resetting the gateway its IP address is displayed in the Client Info column in the BootP/TFTP configuration utility (refer to Figure C-1 on page 351). Use a serial communication software (refer to Section 4.2.4 on page 44). Contact your System Administrator.
Using NOTIFY messages according to <draft-mahy-sipping-signaled-digits-01.txt>: In this mode DTMF digits are carried to the remote side using NOTIFY messages. To enable this mode set: RxDTMFOption = 0 (Declare RFC 2833 in SDP = No) TxDTMFOption = 2 (1st to 5th DTMF Option = NOTIFY) Note that in this mode DTMF digits are erased from the audio stream (DTMFTransportType is automatically set to 0 (DTMF Mute)). Using RFC 2833 relay with Payload type negotiation: In this mode, DTMF digits are carried to the remote side as part of the RTP stream in accordance with RFC 2833 standard. To enable this mode set: TxDTMFOption = 4 (1st to 5th DTMF Option = RFC 2833) RxDTMFOption = 3 (Declare RFC 2833 in SDP = Yes) Note that to set the RFC 2833 payload type with a different value (other than its default, 96) configure the RFC2833PayloadType (RFC 2833 Payload Type) parameter. The gateway negotiates the RFC 2833 payload type using local and remote SDP and sends packets using the PT from the received SDP. The gateway expects to receive RFC 2833 packets with the same PT as configured by the RFC2833PayloadType parameter. If the remote side doesnt include telephony-event in its SDP, the gateway sends DTMF digits in transparent mode (as part of the voice stream). Sending DTMF digits (in RTP packets) as part of the audio stream (DTMF Relay is disabled): Note that this method is normally used with G.711 coders; with other LBR coders the quality of the DTMF digits is reduced. To ser this mode: TxDTMFOption = 0 (1st to 5th DTMF Option = Disable) RxDTMFOption = 0 (Declare RFC 2833 in SDP = No) DTMFTransportType = 2 (DTMF Transport Type = Transparent DTMF)
Notes: The gateway is always ready to receive DTMF packets over IP, in all possible transport modes: INFO messages, NOTIFY and RFC 2833 (in proper payload type) or as part of the audio stream. To exclude RFC 2833 Telephony event parameter from the gateways SDP, set RxDTMFOption = 0 in the ini file.
The following parameters affect the way the MediaPack SIP handles the DTMF digits: TxDTMFOption, RxDTMFOption and RFC2833PayloadType (described in Table 5-7). MGCPDTMFDetectionPoint, DTMFDigitLength and DTMFInterDigitInterval (Table 5-47). DTMFVolume and DTMFTransportType (Table 5-42).
Fax & Modem Transport Modes
Fax/Modem Settings
Users may choose to use one of the following transport methods for fax and for each modem type (V.22/V.23/Bell/V.32/V.34): Fax relay: demodulation / modulation Bypass: using a high bit rate coder to pass the signal Transparent: passing the signal in the current voice coder When the fax relay mode is enabled, distinction between fax and modem is not immediately possible at the beginning of a session. The channel is therefore in Answer Tone mode until a distinction is determined. The packets being sent to the network at this stage are T.38-compliant fax relay packets.
acBoardTemperatureAlarm acBoardEvResettingBoard acBoardEvBoardStarted acgwAdminStateChange acOperationalStateChange
acBoardCallResourcesAlarm acBoardControllerFailureAlarm acFeatureKeyError acBoardOverloadAlarm acActiveAlarmTableOverflow acKeepAlive
acNATTraversalAlarm
acEnhancedBITStatus
acBoardEthernetLinkAlarm acPerformanceMonitoringThresholdCrossi ng
acHTTPDownloadResult
In addition to the listed traps, the device also supports the following standard traps: coldStart authenticationFailure linkDown linkup entConfigChange
SNMP Interface Details
This section describes details of the SNMP interface that is required when developing an Element Manager (EM) for any of the TrunkPack-VoP Series products, or to manage a device with a MIB browser. The gateway offers the following SNMP security features: SNMPv2c community strings SNMPv3 User-based Security Model (USM) users SNMP encoded over IPSec (refer to Section 12.1 on page 279) Combinations of the above Currently, both SNMP and ini file commands and downloads are not encrypted. For ini file encoding, refer to Section E.1.2 on page 365.
14.8.1 SNMP Community Names
By default, the device uses a single, read-only community string of public and a single read-write community string of private. The following community strings can be defined: Up to five read-only community strings Up to five read-write community strings A single trap community string Each community string must be associated with one of the following predefined SNMP groups: Table 14-2: SNMP Predefined Groups Group ReadGroup ReadWriteGroup TrapGroup Gets Access Yes Yes No No Yes No Sets Access Yes Yes Yes Sends Traps
14.8.1.1 Configuration of Community Strings via the Web
For detailed information on configuration the community strings via the Embedded Web Server, refer to Section 5.6.6.2 on page 180.
14.8.1.2 Configuration of Community Strings via the ini File
The following ini file parameters are used to configure community strings: SNMPReadOnlyCommunityString_<x> = '#######' SNMPReadWriteCommunityString_<x> = '#######' Where <x> is a number from 0 to 4. The '#' character represents any alphanumeric character. The maximum length of the string is 20 characters. Version 5.December 2006
The example above creates the following three v3 users: The user "v3user" is defined for a security level of noAuthNoPriv(1) and is associated with ReadGroup1. The user "v3admin1" is defined for a security level of authNoPriv(2) with authentication protocol MD5. The authentication text password is myauthkey and the user will be associated with ReadWriteGroup2. The user "v3admin2" is defined for a security level of authPriv(3) with authentication protocol SHA-1 and privacy protocol DES. The authentication text password is myauthkey, the privacy text password is myprivkey, and the user will be associated with ReadWriteGroup3.
14.8.2.2 Configuring SNMP v3 Users via SNMP
To configure SNMP v3 users, the EM must use the standard snmpUsmMIB and the snmpVacmMIB.
To add a read-only, noAuthNoPriv SNMPv3 user (v3user), take these 3 steps:
1. 2. 9. Clone the row with the same security level. After the clone step, the status of the row is notReady(3). Activate the row (i.e., set the row status to active(1)). Add a row to the vacmSecurityToGroupTable for SecurityName v3user, GroupName ReadGroup1, and SecurityModel usm(3).
A row with the same security level (noAuthNoPriv) must already exist in the usmUserTable. (See the usmUserTable for details).
To delete the read-only, noAuthNoPriv SNMPv3 user (v3user), take these 3 steps:
1. 2. 3. If v3 user is associated with a trap destination, follow the procedure for associating a different user to that trap destination. (See below.) Delete the vacmSecurityToGroupTable row for SecurityName v3user, GroupName ReadGroup1, and SecurityModel usm. Delete the row in the usmUserTable for v3user.
To add a read-write, authPriv SNMPv3 user (v3user), take these 4 steps:
1. 2. 3. 4. Clone the row with the same security level. Change the authentication key and privacy key. Activate the row. That is, set the row status to active(1). Add a row to the vacmSecurityToGroupTable for SecurityName v3admin1, GroupName ReadWriteGroup3, and SecurityModel usm(3).
A row with the same security level (authPriv) must already exist in the usmUserTable (see the usmUserTable for details).

IntraCom Systems, 20337 Via Medici Porter Ranch, CA 91326 Sales: (818) 357-2347 Support: support@intracomsystem.com
AudioCodes MP-114/118 Installation Guide
DEVICE INSTALLATION
Connect the device to the LAN and connect the phone lines. Begin by determining the device IP Address by either of the following methods: a) If the device is using the factory defaults, change the IP address of your computer to 10.1.10.1 with a subnet of 255.255.0.0 so that it can communicate with the device using its factory default IP address of 10.1.10.11. b) If the device is not using the factory defaults, access your routers web interface and view the list of attached devices. The MP-114/118 does not have a Device Name however the displayed Mac Address can be matched to the MAC address printed on the bottom of the device. Use Internet Explorer to access the MP-114/118 web interface configuration pages using the above determined IP address. When prompted enter the default case sensitive user name of Admin and password of Admin. From the displayed home page, observe the firmware version. If the firmware version is prior to version 5.60A.009.005 we recommend upgrading the firmware to the latest version as described in the following Device Firmware Update section.
DEVICE FIRMWARE UPDATE
If the firmware version is prior to version 5.60A.009.005 follow the guidelines to update the firmware version. Begin the upgrade process by downloading the firmware file from the following IntraCom website URL: www.intracomsystem.com/downloads/MP118_SIP_F5.60A.009.005.zip Extract the CMP file from the ZIP to a known location. Use Internet Explorer to access the MP-114/118 web interface configuration pages using the IP address determine in Device Installation. When prompted enter the default case sensitive user name of Admin and password of Admin. ____________________________________________________________________________________
Updated: 4/17/2009 Source: AudioCodes_MP-114-118_Installation_Guide.doc
Locate the Software Update Wizard in your version and start the software upgrade. The method to access to and the behavior of the Software Upgrade Wizard may differ slightly for various firmware versions. When prompted, point the browser to the CMP file extracted previously. Follow all remaining prompts to accept the default settings to complete the firmware upgrade.
VCOM CONFIGURATION
Using the VCOM System Administration application assign a new client for each phone line that will be connected using Client Type of SIP Device: PSTN Telephone Interface (FX0) assigning an appropriate Login name, Login Password and Selector Talk/Listen Name. Assign the selector(s) to at least one Control Panel for testing. Further configuration may be necessary depending on specific functionality required however the default settings will allow for both dial-out and dial-in. For further configuration details please refer to the VCOM System Administration User Guide.
DEVICE CONFIGURATION
Use Internet Explorer to access the MP-114/118 web interface configuration pages using the IP address determine in Device Installation. When prompted enter the default case sensitive user name of Admin and password of Admin. If the IP address is still the factory default or not the desired IP address changes the IP as follows: Under Network Settings -> IP Settings -> Single IP Settings
IP Address: Subnet Mask: Default Gateway Address: <Static IP Address compatible with LAN> 255.255.255.0 <Gateway Address compatible with LAN>
If the IP address of the MP-114/118 was changed and you previously changed the IP address of your computer to match, again change the IP address of your computer to match the new IP address. If an NAT router is between the MP-114/118 and the VCOM server, specific configuration of the MP-114/118 is necessary to properly support NAT router traversal. To do so requires a static public IP address as its STUN server implementation cannot specify a port and therefore cannot handle symmetric NATs. As such the AudioCodes must be configured manually with the NAT routers public IP address as follows: ____________________________________________________________________________________
Network Settings-> IP Settings->NAT Settings
NAT IP Address <NAT Router Public IP Address>
Configure the SIP Server and Proxy settings as follows: Protocol Configuration -> Protocol Definition -> SIP General Parameters
SIP Destination Port
Use Default Proxy Proxy Table Set Proxy Address <VCOM Virtual Matrix SIP IP PORT> [Default is 5060]
Protocol Configuration -> Protocol Definition ->Proxy & Registration
Yes <VCOM Virtual Matrix IP Address> [ NOTE: If the VCOM Virtual Matrix SIP IP Port has been changed from the default of 5060 the address must include the IP Port (eg 208.109.209.117:5061) ] Enable <VCOM Virtual Matrix IP Address or Domain Name> 70 <VCOM Virtual Matrix IP Address or Domain Name>
Enable Registration Registrar Name: Registration Time: Gateway Name:
Configure the Audio Codec settings as follows: Protocol Configuration -> Protocol Definition -> Coders
Coder Name G.711A-Law G.711U-Law Packet Time Rate Payload Type Silence Suppression Yes Yes
Configure the SIP Client settings as follows: Protocol Configuration -> Endpoint Settings -> Authentication
Gateway Port Port 1 FXO Port 2 FXO Port 3 FXO Port 4 FXO Gateway Port Port 1 FXO Port 2 FXO Port 3 FXO Port 4 FXO User Name <VCOM Login Name for Line 1> <VCOM Login Name for Line 2> <VCOM Login Name for Line 3> <VCOM Login Name for Line 4> Password <VCOM Login Password for Line 1> <VCOM Login Password for Line 2> <VCOM Login Password for Line 3> <VCOM Login Password for Line 4> Auto Dial Status Enable Enable Enable Enable
Protocol Configuration -> Endpoint Settings -> Automatic Dialing
Destination Phone Number *<VCOM Login Name for Line 1> *<VCOM Login Name for Line 2> *<VCOM Login Name for Line 3> *<VCOM Login Name for Line 4>
[Asterisk is required] [Asterisk is required] [Asterisk is required] [Asterisk is required]
____________________________________________________________________________________
Protocol Configuration -> Endpoint Settings -> Caller Display Information
Gateway Port Port 1 FXO Port 2 FXO Port 3 FXO Port 4 FXO Caller ID / Name Anonymous Anonymous Anonymous Anonymous Caller ID Enabled Enabled Enabled Enabled Phone Number Hunt Group Id <VCOM Login Name for Line 1> <VCOM Login Name for Line 2> <VCOM Login Name for Line 3> <VCOM Login Name for Line 4> Tel Profile ID Presentation Allowed Allowed Allowed Allowed
Protocol Configuration -> Endpoint Settings -> Caller ID Permissions
Gateway Port Port 1 FXO Port 2 FXO Port 3 FXO Port 4 FXO Channel(s) 3 4
Protocol Configuration -> Endpoint Number -> Endpoint Phone Number
Codec Configuration
Configure the general devices settings as follows: Advanced Applications -> FXO Settings
Dialing Mode Wait for Dial Tone Two Stages Yes
Optionally configure the general devices settings as follows:
With some PSTN networks it has been observed that the call may not disconnect automatically on remote user disconnection. The following parameters can be modified which may or may not resolve the problem. Protocol Configuration -> SIP Advanced Parameters -> Advanced Parameters
Enable Polarity Reversal Enable (? Required?)
Enable Current Disconnect Enable (? Required?)
An additional option which may help resolve the issue has no web GUI user interface. To modify this setting Load and edit the configuration INI file to include the following
[Analog Params] CurrentDisconnectDuration = 200
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