Copyright and Legal Notice

Copyright © 2007-2010 Dialogic Corporation. All Rights Reserved. You may not reproduce this document in whole or in part without permission in writing from Dialogic Corporation at the address provided below.

All contents of this document are furnished for informational use only and are subject to change without notice and do not represent a commitment on the part of Dialogic Corporation or its subsidiaries ("Dialogic"). Reasonable effort is made to ensure the accuracy of the information contained in the document. However, Dialogic does not warrant the accuracy of this information and cannot accept responsibility for errors, inaccuracies or omissions that may be contained in this document.

INFORMATION IN THIS DOCUMENT IS PROVIDED IN CONNECTION WITH DIALOGIC® PRODUCTS. NO LICENSE, EXPRESS OR IMPLIED, BY ESTOPPEL OR OTHERWISE, TO ANY INTELLECTUAL PROPERTY RIGHTS IS GRANTED BY THIS DOCUMENT. EXCEPT AS PROVIDED IN A SIGNED AGREEMENT BETWEEN YOU AND DIALOGIC, DIALOGIC ASSUMES NO LIABILITY WHATSOEVER, AND DIALOGIC DISCLAIMS ANY EXPRESS OR IMPLIED WARRANTY, RELATING TO SALE AND/OR USE OF DIALOGIC PRODUCTS INCLUDING LIABILITY OR WARRANTIES RELATING TO FITNESS FOR A PARTICULAR PURPOSE, MERCHANTABILITY, OR INFRINGEMENT OF ANY INTELLECTUAL PROPERTY RIGHT OF A THIRD PARTY.

Dialogic products are not intended for use in medical, life saving, life sustaining, critical control or safety systems, or in nuclear facility applications.

Due to differing national regulations and approval requirements, certain Dialogic products may be suitable for use only in specific countries, and thus may not function properly in other countries. You are responsible for ensuring that your use of such products occurs only in the countries where such use is suitable. For information on specific products, contact Dialogic Corporation at the address indicated below or on the web at www.dialogic.com.

It is possible that the use or implementation of any one of the concepts, applications, or ideas described in this document, in marketing collateral produced by or on web pages maintained by Dialogic may infringe one or more patents or other intellectual property rights owned by third parties. Dialogic does not provide any intellectual property licenses with the sale of Dialogic products other than a license to use such product in accordance with intellectual property owned or validly licensed by Dialogic and no such licenses are provided except pursuant to a signed agreement with Dialogic. More detailed information about such intellectual property is available from Dialogic's legal department at 9800 Cavendish Blvd., 5th Floor, Montreal, Quebec, Canada H4M 2V9.

Dialogic encourages all users of its products to procure all necessary intellectual property licenses required to implement any concepts or applications and does not condone or encourage any intellectual property infringement and disclaims any responsibility related thereto. These intellectual property licenses may differ from country to country and it is the responsibility of those who develop the concepts or applications to be aware of and comply with different national license requirements.

Any use case(s) shown and/or described herein represent one or more examples of the various ways, scenarios or environments in which Dialogic^® products can be used. Such use case(s) are non-limiting and do not represent recommendations of Dialogic as to whether or how to use Dialogic products.

Dialogic, Dialogic Pro, Brooktrout, Diva, Diva ISDN, Making Innovation Thrive, Video is the New Voice, Diastar, Cantata, TruFax, SwitchKit, SnowShore, Eicon, Eicon Networks, NMS Communications, NMS (stylized), Eiconcard, SIPcontrol, TrustedVideo, Exnet, EXS, Connecting to Growth, Fusion, Vision, PacketMedia, NaturalAccess, NaturalCallControl, NaturalConference, NaturalFax and Shiva, among others as well as related logos, are either registered trademarks or trademarks of Dialogic Corporation or its subsidiaries. Dialogic's trademarks may be used publicly only with permission from Dialogic. Such permission may only be granted by Dialogic's legal department at 9800 Cavendish Blvd., 5th Floor, Montreal, Quebec, Canada H4M 2V9. Any authorized use of Dialogic's trademarks will be subject to full respect of the trademark guidelines published by Dialogic from time to time and any use of Dialogic's trademarks requires proper acknowledgement.

The names of actual companies and products mentioned herein are the trademarks of their respective owners.

This document discusses one or more open source products, systems and/or releases. Dialogic is not responsible for your decision to use open source in connection with Dialogic products (including without limitation those referred to herein), nor is Dialogic responsible for any present or future effects such usage might have, including without limitation effects on your products, your business, or your intellectual property rights.

Software License Agreement

This Software License Agreement ("Agreement") is a legal Agreement for the conditional use of the software provided herein ("Program") between you, the end-user, whether an individual or a company and all of its authorized users of the Program ("You"), and Dialogic Corporation ("Dialogic"). By downloading, copying or otherwise using the Program, You agree to be bound by the terms of the foregoing license.

YOU ASSUME RESPONSIBILITY FOR THE SELECTION OF THE PROGRAM TO ACHIEVE YOUR INTENDED RESULTS, AND FOR THE INSTALLATION, USE, AND RESULTS OBTAINED FROM THE PROGRAM.

The Program may contain certain software applications and portions of applications which are provided to You under terms and conditions which are different from this Agreement (such as open source or community source), or which require Dialogic to provide Customer and its Subsidiaries with certain notices and/or information ("Excluded Code"). Dialogic will identify such Excluded Code in a text file or about box or in a file or files referenced thereby (and shall include any associated license agreement, notices and other related information therein), or the Excluded Code will contain or be accompanied by its own license agreement. Your use of the Excluded Code will be subject to the terms and conditions of such other license agreement solely to the extent such terms and conditions are inconsistent with the terms and conditions of this Agreement or are required by such other license agreement. By using or not uninstalling such Excluded Code after the initial installation of the Excluded Code You acknowledge and agree to all such license agreements, notices and information.

License

Under the terms and conditions of this Agreement:

• You may install and use one copy of the Program on a single-user computer, file server, or on a workstation of a local area network.
• Some or all functions of the Program may be available solely if the Program is used with one or more legally acquired Dialogic Activation Key(s).
• To obtain an Activation Key You must first purchase a Proof of Purchase Code (PPC). A PPC may be included in Your software or hardware package or You may have to purchase it separately.
• You will receive Your Activation Key upon registering the Proof of Purchase Code as directed in the PPC document.
• It may be possible to install multiple Activation Keys into the Program; in such a case, the total functionality provided by the Program will be the sum of the licensed functionalities controlled by the installed Activation Keys as long as the maximum capabilities of the Program are not exceeded and the functionalities are compatible.
• Your Activation Key(s) will restrict Your use of the Program. At least one of the following restriction schemes will be available to You when You register each PPC and request an Activation Key.
  • The Activation Key may be associated with a specific Dialogic hardware device. In this case, the licensed functionality controlled by the Activation Key will be available solely if the same Dialogic hardware is present in the computer. You can move the Program to another computer solely if You move the specified Dialogic hardware to the new computer.
  • The Activation Key may be associated with a specific Dialogic-supplied software protection device ('dongle'). In this case, the licensed functionality controlled by the Activation Key will be available solely if the same dongle is present in the computer. You can move the Program to another computer solely if You also move the dongle to the new computer.
  • The Activation Key may be associated with Your specific computer hardware platform. In this case, the licensed functionality controlled by the Activation Key will be available solely if no significant change is made to the hardware installed in the computer. Replacement Activation Keys may be issued at the discretion of Dialogic solely if Dialogic can determine that You have not moved the Program to another computer. Sufficient information must be provided to Dialogic to allow it to make that determination.
• In addition to the above restrictions, each Activation Key may have a specific term of use commencing from the date of PPC registration. In this case, the licensed functionality controlled by the Activation Key will not be available after the Activation Key has expired.
• The Activation process requires that You enter the following information into the web-based system to obtain an Activation Key:
  • PPC
  • The Device ID provided to You by the "Activation" function in the Program
  • Your email address so that the Activation Key can be delivered to You by email
• Dialogic will retain the information above for the following purposes:
  • Validation of future requests from You for replacement Activation Keys
  • Sending renewal reminders to You in the case of limited time licenses.
• If the Dialogic hardware device that Activation Keys are associated with is judged to be defective by Dialogic following its standard practices, Dialogic's Support department will issue to You replacement Activation Keys associated with the replacement device upon receipt of the faulty device by Dialogic. Replacement of the faulty device is subject to the terms of Dialogic's standard Hardware product Warranty in effect at the time You purchased the hardware product concerned ("Hardware Warranty"). If a valid Advance Replacement Insurance policy contract is in place for the Dialogic hardware product concerned, Dialogic will endeavor to expedite provision of Activation Keys associated with the replacement device.
• If the Dialogic-supplied software protection device (a "dongle" or "USB-stick") that Activation Keys are associated with is judged to be defective by Dialogic following its standard practices, Dialogic Support will issue to You replacement Activation Keys associated with the replacement device upon receipt by Dialogic of the failed device. Replacement of the faulty device is subject to the terms of Dialogic's Hardware Warranty.
• You may copy the Program into any machine readable or printed form for backup in support of Your use of one copy of the Program;
• You may make one copy of Dialogic's documentation provided that all copyright notices contained within the documentation are retained;
• You may transfer the Program, documentation and the license to another eligible party within Your Company if the other party agrees to accept the terms and conditions of this Agreement. If You transfer the Program and documentation You must at the same time either transfer all copies whether in printed or machine readable form to the same party or destroy any copies not transferred;
• You may not rent or lease the Program. You may not reverse engineer, decompile or disassemble the Program. You may not use, copy, modify or transfer the Program and documentation, or any copy, modification or merged portion, in whole or in part, except as expressly provided for in this Agreement;
• You may not modify the Program in order to circumvent or subvert the protection mechanisms inherent in the program or attempt to use a time-limited Activation Key after it has expired;
• If You transfer possession of any copy, modification or merged portion of the Program or documentation to another party in any way other than as expressly permitted in this Agreement, this license is automatically terminated.

Term

The license is effective until terminated. You may terminate it at any time by destroying the Program and documentation together with all copies, modifications and merged portions in any form.

It will also terminate upon conditions set forth elsewhere in this Agreement or if You fail to comply with any terms or conditions of this Agreement at any time. You agree upon such termination to destroy the Program and documentation together with all copies, modifications and merged portions in any form.

Limited Warranty

The only warranty Dialogic makes, beyond the replacement of Activation Keys under the terms set out above, is that the medium on which the Program is recorded will be replaced without charge if Dialogic, in good faith, determines that it was defective in materials or workmanship and if returned to Your supplier with a copy of Your receipt within ninety (90) days from the date You received it. Dialogic offers no warranty for Your reproduction of the Program. This Limited Warranty is void if failure of the Program has resulted from accident, misuse, abuse or misapplication.

Customer Remedies

Dialogic's entire liability and Your and Your Authorized Users exclusive remedy shall be, at Dialogic's option, either (a) return of the price paid or (b) repair or replacement of the Program that does not meet the above Limited Warranty. Any replacement Program will be warranted for the remainder of the original Warranty period.

No Other Warranties

Dialogic disclaims all other warranties, either express or implied, including but not limited to implied warranties or merchantability and fitness for a particular purpose and the warranty against latent defects, with respect to the Program and the accompanying documentation. This limited warranty gives You specific legal rights. You may have others, which may vary from jurisdiction to jurisdiction.

No Liability for Consequential Damage

In no event shall Dialogic, its affiliates or its suppliers or their officers, directors or employees be liable for any damages whatsoever (including without limitation, damages for loss of business profits, business interruption, loss of information, or other pecuniary loss and indirect, consequential, incidental, economic or punitive damages) arising out of this Agreement or the use of or inability to use this Program, even if Dialogic has been advised of the possibility of such damages. As some jurisdictions do not allow the exclusion or limitation for certain damages, some of the above limitations may not apply to You or Your Authorized Users.

Limit of Liability

Dialogic's entire aggregate liability under any provision of this Agreement shall be limited to the amount actually paid by You for the affected Program.

Right to Audit

If this Program is licensed for use in a Company, Your Company agrees to keep all usual and proper records and books of accounts and all usual proper entries relating to each installation of the Program during the term of this Agreement and for a period of three (3) years thereafter. During this period, Dialogic may cause an audit to be made of the applicable records and of the installations of the Program in order to verify Your compliance with this Agreement and prompt adjustment shall be made to compensate for any errors or omissions disclosed by such audit. Any such audit shall be conducted by an independent certified public accountant selected by Dialogic and shall be conducted during the regular business hours at Your offices and in such a manner as not to interfere with Your normal business activities. Any such audit shall be paid for by Dialogic unless material discrepancies are disclosed.

For such purposes, 'material discrepancies' shall mean the Company exceeding by three percent (3%) or more the number of licensed channels for any function of the Program or the Company exceeding the licensed number of Authorized Users by three percent (3%) or more. If material discrepancies are disclosed, Your Company agrees to pay Dialogic for the costs associated with the audit as well as the license fees for the additional licensed channels or additional Authorized Users. In no event shall audits be made more frequently than semi-annually unless the immediately preceding audit disclosed a material discrepancy.

Supplementary Software

Any Supplementary Software provided with the Dialogic Program referred to in this License Agreement is provided "as is" with no warranty of any kind.

U.S. Government Restricted Rights

The Program and documentation are provided with RESTRICTED RIGHTS. Use, duplication or disclosure by the U.S. Governments is subject to restrictions as set forth in subparagraph c) 1) ii) of The Rights in Technical Data and Computer Software clause at DFARS 252.227-7013 or subparagraph c) 1) and 2) of the Commercial Computer Software-Restricted Rights at 48 CFR52.227-19, as applicable.

Governing Law

This Agreement shall be construed and controlled by the laws in force in the Province of Quebec, Canada with the exception of its conflict of laws rules. The United Nations Convention on Contracts for the Sale of Goods does not apply to this Agreement.

Miscellaneous

All rights and remedies, whether conferred hereunder or by any other instrument or law, will be cumulative and may be exercised singularly or concurrently. Failure by either Dialogic or You to enforce any term will not be deemed a waiver of future enforcement of that or any other term. The terms and conditions stated herein are declared to be severable. Should any term(s) or condition(s) of this Agreement be held to be invalid or unenforceable the validity, construction and enforceability of the remaining terms and conditions of this Agreement shall not be affected. You may not assign any rights, duties or obligations hereunder without the prior written approval of Dialogic and any attempt to assign any rights, duties or obligations hereunder without Dialogic's written consent will be void. It is expressly agreed that Dialogic and You are acting as independent contractors. These terms and conditions constitute the entire agreement between Dialogic and You with respect to the subject matter hereof. These terms and conditions will prevail notwithstanding any different, conflicting or additional terms and conditions that may appear on any other agreement between Dialogic and You. Deviations from these terms and conditions are not valid unless agreed to in writing in advance by an authorized representative of Dialogic. Any notice to Dialogic under this Agreement must be sent by courier or registered mail to the attention of Dialogic's legal department at the address below.

Contractor/ manufacturer is:

DIALOGIC CORPORATION.

9800 Cavendish Blvd., Montreal, Quebec, Canada H4M 2V9

This Agreement has been drafted in English at the express wish of the parties. Ce contrat a été rédigé en anglais à la demande expresse des parties.

Feature Overview

General Features

• Noise suppression support
• Echo cancellation selectable via GUI
• Diva SIPcontrol software update via web interface
• Interoperability with Dialogic^® Host Media Processing (HMP) software 3.0WIN and 3.1LIN
• Configuration via web interface
• Standard web browsers can be used for configuring. The Dialogic^® Diva^® SIPcontrol^TM Software has been tested with Mozilla Firefox version 2.0 and 3.0
• Remote configuration of the Diva SIPcontrol software from any computer in the network
• Cause codes: Configurable translation of ISDN cause code to SIP response code and vice versa; consequently, the Diva SIPcontrol software can adapt to the specific behavior of the PSTN, PBX, and/or SIP peer.
• Support for North-American numbering plan: The configuration of multiple area codes are handled as local. Therefore, the Dialogic^® Diva^® SIPcontrol^TM Software dialplan engine is able to automatically format dialed numbers according to local phone provider requirements without any additional regular expressions.
• Configuration changes during runtime: Modify most parameters of the Diva SIPcontrol software without the need to restart the service; active calls are not affected by configuration updates and continue undisturbed.
• Interoperability with the Dialogic^® Brooktrout^® Bfv API SDK: The Dialogic^® Brooktrout^® SR140 Fax Software version 5.2.1 has been confirmed via testing to be V.34/T.38 interoperable with the Diva SIPcontrol software. The Brooktrout SR140 Fax Software is high-performance, host-based T.38 fax software for IP networks.
• Codec Configuration: Configuration options for supported audio and fax codecs. See Media processing for the supported codecs.
• Support for Proxy and Registrar registration
• Support for registering the Diva SIPcontrol software as an e-phone gateway
• Support for Early Media: Early Media is supported for calls from SIP to the PSTN. For calls from the PSTN to SIP, it depends on the used line protocol.
• Configuration of Dialogic^® Diva^® Media Board parameters via the web interface
• Interoperability with the Dialogic^® Host Media Processing (HMP) software
• Support for up to 64 ports per system for Dialogic^® Diva^® BRI and Analog Media Board installations
• Support for up to 240 ports per system for Dialogic^® Diva^® PRI Media Board installations

Call handling

• SIP methods: ACK, BYE, INVITE, NOTIFY*, REFER, CANCEL, OPTIONS, PRACK
• Configurable IP transport layer TCP, UDP, or TLS
• Support for TLS and SSL encryption and authentication
• Support for SRTP (secure Real-time Transport Protocol)
• Support for SIPS (Secure SIP)
• Basic call including numbering services:
  • Called Party Number
  • Calling Party Number
  • Redirecting Number
• Call Routing
• Call Hold/Retrieve (e.g., Re-Invite mapping towards ISDN)
• PSTN-side Call Transfer (i.e., Refer points to PSTN)
• PSTN-side incoming Call Diversion
• Message Waiting Activation / Deactivation
• Support of REDIRECT (Moved Temporarily)
• SIP Session Timer (RFC 4028)
• Simplified number normalization based on PSTN connection parameters
• Number manipulation using regular expressions

* NOTIFY, in combination with SUBSCRIBE, is used to provide the feature Message Waiting Activation / Deactivation with regular SIP clients. However, in a gateway configuration, applications use the features without the need for the Diva SIPcontrol software to use SUBSCRIBE.

Media processing

• Support for the following codecs:
  • G.711 A-law and µ-law
  • G.726 (16,24,32, and 40 kbps)
  • G.729
  • GSM-FR
  • iLBC

Note: For G.729, you need to purchase and activate a license before you can use it. See License Activation for more information.

Note: iLBC is only available on Dialogic^® Diva^® Multiport Media Boards, on Dialogic^® Diva^® V-4PRI PCIe HS Media Boards

• RTP dynamic payload audio/telephony event
• RTP profile RTP/AVP
• DTMF via RTP payload/telephony event (RFC 2833)
• PSTN-side fax tone detection via RTP event (RFC 2833)
• Echo Canceller with 128 ms echo tail used with RTP (On some Diva Media Boards up to 256 ms. See the Dialogic^® Diva^® System Release Reference Guide for more information.)
• T.38 fax up to V.34 (SuperG3 Fax)

Reliability

• Load balancing and failover on PSTN and on SIP side (optionally uses OPTIONS for Keep-Alive check)
• Alive check for active calls on SIP side via SIP session timer

Supported RFCs

• RFC2617 - HTTP Digest Authentication
• RFC2833 - RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals
• RFC3261 - Session Initiation Protocol
• RFC3262 - Reliability of Provisional Responses in Session Initiation Protocol (SIP)
• RFC3264 - An Offer/Answer Model with Session Description Protocol
• RFC3265 - SIP-specific Event Notification
• RFC3326 - The Reason Header Field for the Session Initiation Protocol (SIP)
• RFC3389 - RTP Payload for Comfort Noise
• RFC3398 - ISDN to SIP mapping
• RFC3420 - Internet Media Type message/sipfrag
• RFC3515 - REFER method
• RFC3550 - Realtime Transport Protocol (RTP)
• RFC 3551 - RTP/AVP profile
• RFC3711 - The Secure Real-time Transport Protocol (SRTP)
• RFC3842 - Message Waiting Indication for SIP
• RFC3891 - SIP "Replaces" header
• RFC3892 - SIP Referred - By Mechanism
• RFC3851 - Internet Low Bit Rate Codec (iLBC)
• RFC3952 - Real-time Transport Protocol (RTP) Payload Format for internet Low Bit Rate Codec (iLBC) Speech
• RFC3960 - Early Media and Ringing Tone Generation in the Session Initiation Protocol (SIP), only gateway model
• RFC4028 - Session Timers in SIP
• RFC4497 - Interworking between SIP and QSIG
• RFC4566 - Session Description Protocol (SDP)
• RFC4568 - SDP Security for Media Streams
• Draft: Diversion Indication in SIP (draft-levy-sip-diversion-08)

Enhanced routing

• Defines which CAPI controller is used for which calls from SIP
• Increased flexibility of load balancing and failover functionality; load-balancing and failover can be used together and are available for calls to the PSTN as well.
• Number-based routing also available for calls to the PSTN
• Matching rules for number-based routing can contain regular expressions
• Routing based on calling or redirected number, the redirected number is only available for calls from the PSTN

Enhanced address manipulation

• Number manipulation is defined on three different stages of the call routing (inbound, route selection, outbound).
• Unlimited number of regular expressions for number manipulation at each stage of call routing
• Different dialplans can be entered for each controller and each SIP peer, which can ease the deployment in an environment with multiple locations.

Supported Software

The Dialogic^® Diva^® SIPcontrol^TM software requires Dialogic^® Diva^® System Release 9.5LIN software.

Supported Operating Systems

The Dialogic® Diva® SIPcontrolTM software supports Linux 32-bit and 64-bit. For supported kernel versions, see the Dialogic® Diva® System Release LIN Reference Guide.

Supported Hardware

The Dialogic^® Diva^® SIPcontrol^TM Software supports the following Dialogic^® Diva^® Media Boards (up to 240 channels):

Dialogic^® Diva^® BRI Media Boards

Dialogic^® Diva^® PRI Media Boards

Note: "HS" stands for the half size and "FS" for the full size board format.

Dialogic^® Diva^® Analog Media Boards

¹⁾ After the installation, the Dialogic^® Diva^® V-BRI and V-Analog Media Boards are displayed as Dialogic^® Diva^® UM-BRI and UM-Analog Media Boards.

Software Installation

The Dialogic^® Diva^® SIPcontrol^TM software is automatically installed together with the Dialogic^® Diva^® for Linux package in /usr/lib/opendiva/diva.sipcontrol.

License Activation

The Dialogic^® Diva^® SIPcontrol^TM Software includes a default license for two channels. This license can be used for testing and evaluating the Diva SIPcontrol software.

You must activate a license if you need more than the two channels of the default license included with the Diva SIPcontrol software, or if you want to use G.729 speech compression, V.17 fax, or V.34 fax offered with the installed Dialogic^® Diva^® Media Board. During the activation process of the license, you need to choose a Diva Media Board to which the license should be bound. After having activated the license for this Diva board, the license cannot be transferred to be used with another Diva board.

Notes:

• Licenses for G.729, V.17, and V.34 need to be activated in the Dialogic^® Diva^® Configuration Manager. See the Dialogic^® Diva^® Configuration Manager Online Help for more information.
• The fax functionality needs to be licensed only for Dialogic^® Diva^® PRI Media Boards with multiple ports.
• Dialogic^® Host Media Processing (HMP) Software licenses for SIP channels are also valid for SIPcontrol, but they require HMP software to be installed on the same system as the Diva SIPcontrol software.

To activate your license file, you need the Device Unique ID (DUID) and the Proof of Purchase Code (PPC).

Once you have both, the DUID and the PPC, visit the Dialogic activation web site to register your PPC together with the DUID to receive the license file. See To register your DUID and PPC for more information. Activate this license file in the Diva SIPcontrol software configuration web interface. For more information, see To activate the license file.

Device Unique ID (DUID)

The DUID binds the installed Dialogic^® Diva^® SIPcontrol^TM Software to your PC (PC fingerprint).

To get the DUID:

1. Open the Diva SIPcontrol software configuration web interface.
2. Click License Management on the left side of the Diva SIPcontrol software configuration web interface to open the License Status dialog.
3. In the License Status dialog, copy the DUID number of the Dialogic^® Diva^® Media Board you want to activate to the clipboard.
4. If you need to do the web activation using another computer, open an editor, paste the DUID, and save the file.

Proof of Purchase Code (PPC)

When you purchase the Dialogic^® Diva^® SIPcontrol^TM Software license, you receive a PPC either in printed form or via email. By registering this PPC, you represent and warrant that you lawfully purchased the license.

To register your DUID and PPC

Once you have your DUID and PPC, you can register them as follows:

1. Open the following web site: www.dialogic.com/activate.
2. Enter the PPC and click Check.

   

3. If your PPC is valid, the following web site will open:

   

   Paste your Device Unique ID (DUID) that you saved earlier, and enter your email address to which the license file should be sent.

4. Click Activate to generate the license file that will be sent to the email address you have entered.
5. Save the received license file and activate it. For more information, see To activate the license file.

To activate the license file

After you have received your license file by email and have saved it, you can activate it.

Note: The date set in the system settings of your computer must be correct. Otherwise, you cannot add your license file.

To activate the license:

1. Click License Management on the left side of the Dialogic^® Diva^® SIPcontrol^TM Software configuration web interface to open the License Status dialog.
2. In the License Status dialog, click Browse, go to the directory in which you saved the license file and click Open.
3. Click Upload to activate the license file.

Dialogic® Diva® SIPcontrol™ Software Configuration

The Dialogic^® Diva^® SIPcontrol^TM Software can be configured via the Dialogic^® Diva^® SIPcontrol^TM Software web interface.

To open the Diva SIPcontrol software web interface:

1. Open an internet browser on the computer where the Diva SIPcontrol software is installed and enter http://127.0.0.1:10005. By default, the access to the web interface is only allowed from the localhost (127.0.0.1), and by default, the port number to which the server is listening is set to 10005.
2. If you need to access the configuration via remote access, you must set a password. To do so, open the main configuration web interface and click Password on the left hand side under Configuration. Enter an at least 7 digit long password and confirm it. Click Save to make the new password active.
3. For remotely accessing the configuration, open the port in the local firewall settings.
4. Restart the Dialogic^® Diva^® WebConfig service.

Now, you can access the Diva SIPcontrol software web interface on any of the IP addresses of the PC where SIPcontrol is installed. The Diva SIPcontrol software configuration is divided into the following sections:

• PSTN Interfaces
• Network Interfaces
• SIP Peers
• Routings
• Security Profiles
• Dialplans
• Address Maps
• Cause Code Maps
• Codec Profiles
• Registrations
• System Settings

Mandatory configurations are:

1. Choose and enable one network interface.
2. Create and enable one SIP peer.
3. Create and configure one route for PSTN to SIP calls and another route for SIP to PSTN calls.

Configuration Tips and Hints

• Changes to the configuration will only take effect after you clicked Save at the bottom of the main configuration page.
• All settings will be lost if you close the Dialogic^® Diva^® SIPcontrol^TM software web interface without having saved the configuration at the bottom of each configuration page.
• A restart of the Diva SIPcontrol software is recommended if you change the IP address or the port on which SIPcontrol is listening. If you do not restart, the Diva SIPcontrol software will continue listening on the previously configured port and IP address.

  Note: The restart will terminate active connections.

• All names for specific configuration elements are limited to 32 alphanumeric characters and must not be repeated, i.e., you cannot assign the same name for two SIP peers.
• The configuration session times out after 30 minutes of inactivity and a new login is required to access the session again. If the new login screen appears when you try to save the configuration, login again and click the "Back" button of the browser. The configuration session opens with the settings before the time out and you can save the configuration.
• To remove the password login page, logout from the web interface and restart the Dialogic^® Diva^® WebConfig service.
• To use TLS for SIP calls, upload the certificates as described under Security Profiles and enable the TLS port as described under Network Interfaces.
• To open the online help for a specific parameter, click the parameter and a window with the help text will pop up.

PSTN Interfaces

This section describes the Dialogic^® Diva^® SIPcontrol^TM Software's PSTN interface related settings, e.g., which lines are used by the Diva SIPcontrol software or how Call Transfer is performed on this line. Line Parameters such as the signalling protocols (Q.Sig, ETSI) can be configured on the Board Configuration page.

At least one PSTN interface must be enabled for the Diva SIPcontrol software to be able to work. Disabled PSTN interfaces are ignored for both inbound and outbound calls. For each line, you may select a dialplan that you can configure in Dialplan Configuration.

To change the settings for the enabled controller, click Details on the right hand side. To open the online help for a specific parameter, click the parameter and a window with the help text will pop up.

Note: PSTN interfaces without a binding to the CAPI service in the Dialogic^® Diva^® Configuration Manager are disabled in the Diva SIPcontrol software web interface and cannot be configured.

The following configuration menus are available for each Diva Media Board:

• General
• Enhanced
• Address Normalization
• PSTN Call Transfer Settings
• Message Waiting Indication (MWI)

General

You can configure the parameters shown in the graphic and explained below:

Enhanced

Here you may configure the settings for Early media support. You may configure the parameters shown in the graphic and explained below:

Address Normalization

You may configure the parameters shown in the graphic and explained below:

PSTN Call Transfer Settings

Some Call Transfer options can be configured in the Blind Call Transfer section and in the Supervised Call Transfer section.

Message Waiting Indication (MWI)

You may configure the parameters shown in the graphic and explained below:

Network Interfaces

The network interface configuration allows for configuring the global network parameters of the Dialogic^® Diva^® SIPcontrol^TM Software, such as the IP addresses and the ports on which the Diva SIPcontrol software will be listening. The Diva SIPcontrol software supports only a single IP address and port number. To open the online help for a specific parameter, click the parameter and a window with the help text will pop up.

You may configure the parameters shown in the graphic and explained below:

SIP Peers

A SIP peer is a specific endpoint to and from which the Dialogic^® Diva^® SIPcontrol^TM Software will establish calls. The peer-specific settings may be used to adapt the Diva SIPcontrol software's behavior towards this peer.

To add a SIP peer, click Add. To change the settings for the enabled SIP peer, click Details on the right hand side. To open the online help for a specific parameter, click the parameter and a window with the help text will pop up. The following menus are available:

• General
• Enhanced
• Security
• Session Timer
• Address Normalization
• Authentication

General

You may configure the parameters shown in the graphic and explained below:

Enhanced

You may configure the parameters shown in the graphic and explained below:

Security

You may configure the parameters shown in the graphic and explained below:

Session Timer

You may configure the parameters shown in the graphic and explained below:

Address Normalization

You may configure the parameters shown in the graphic and explained below:

Authentication

You may configure the parameters shown in the graphic and explained below:

Routing

The routing configuration defines the destination to which incoming calls are forwarded. Possible criteria that may determine the destination are:

• Called, calling, and redirected number or SIP address of a call for which the redirected number is only available for calls originating in the PSTN,
• the source where a call originated, i.e., a PSTN interface name or a specific SIP peer,
• the current channel allocation across a set of several possible destinations in a load-balancing environment, and
• the current status of a destination. See How a Call Is Processed for more information.

For more information about possible routing configurations, see Routing Examples.

To add a routing, click Add. To configure an existing routing, click Details. Since routes are processed in their configured order, the first matching route takes the call. To change the order, click the "arrow up" and "arrow down" buttons. To open the online help for a specific parameter, click the parameter and a window with the help text will pop up.

Edit Routing Configuration

You may configure the parameters shown in the graphic and explained below:

Address Normalization For Condition Processing (Using Source Dialplan)

You may configure the parameters shown in the graphic and explained below:

Conditions

You may configure certain conditions for a route. If you do not configure any conditions, the route is used as default route.

Note: If prefixes need to match, the digits of the prefix need to be prepended by a caret symbol ("^"), otherwise these digits would match within the number as well, e.g. 0 would also match 1230@sipcontrol.com.

You may configure the parameters shown in the graphic and explained below:

Address Manipulation

You may configure the parameter shown in the graphic and explained below:

Security Profiles

When you use the Transport Layer Security (TLS) protocol for secure communication, you need to set various security settings.

Upload Certificate and Key Files

For authentication and data encryption, certificates need to be installed on the computer with the Diva SIPcontrol software and on remote computers. When a secure domain is opened, server and client authentify each other with a so called "SSL handshake". With this handshake, the identity of a user is certified and the user can be trusted. All necessary certificates are provided by a Certificate Authority (CA) and they are issued for one domain name. For test purposes or internal usage, you can also create and sign your own self-signed certificate, e.g., with one of the many tools available on the internet, just search for "self-signed certificate" and you will find a list of possible tools. But you need to be aware that self-signed certificates do not provide the same security as CA-signed certificates. All files need to be in "pem" format, that means base64 encoded. Also, many web browsers check if the certificate is signed by a CA, and, if it is not, a warning message will pop up asking whether the user really wants to trust that web site, which can make the user feel insecure.

A default certificate is provided with the software, but for security reasons, you should install your own web server certificate.

Certificate files can be generated in different formats, e.g., .pem, .der, .cer, or .pfx. All files need to be in "pem" format, that means base64 encoded, in order to be used by the Diva SIPcontrol software.

Note for CER files: CER files can be renamed to .pem directly if they are base64 encoded. No bag attribute lines and/or additional CR and empty lines are allowed. If CER files are ASN.1 coded, they need to be converted to with a converter tool.

Note for PFX files: The PFX or PKCS#12 format is a binary format for storing the server certificate, any intermediate certificates, and the private key in one encryptable file. When converting a PFX file to PEM format, tools like OpenSSL will put all the certificates and the private key into a single file. You will need to open the file in a text editor and copy each certificate and private key (including the BEGIN/END statments) to its own individual text file and save them as certificate.cer, CACert.cer, and privateKey.key respectively.

How to retrieve keys and certificates from a PFX file for use in the Diva SIPcontrol software

In the following procedure openssl is used as example converter tool.

1. Export the private key file from the PFX file:

   openssl pkcs12 -in filename.pfx -nocerts -out protected-key.pem

2. Remove the passphrase from the private key as required by the Diva SIPcontrol software:

   openssl rsa -in protected-key.pem -out key.pem

3. Export the certificate file from the PFX file:

   openssl pkcs12 -in filename.pfx -clcerts -nokeys -out cert.cer

4. Export the Root CA certificate file from the PFX file:

   openssl pkcs12 -in filename.pfx -cacerts -nokeys -out cacert.cer

For more information about secure connections and certificates, see Data Security Overview.

The screen below shows the web interface with no certificates uploaded:

To upload a certificate:

1. Click Browse, in the File Upload window go to the folder where the certificate file is located, and click Open.
2. In the Diva SIPcontrol software web interface click Upload. After the certificates are uploaded, the information below the certification files changes from "Not available" to "Uploaded". Note that you will not see the paths to the directory in which the files are stored:

Global Security Parameters

Dialplans

With help of the local phone settings, the Dialogic^® Diva^® SIPcontrol^TM Software is able to convert a received call address to a normalized form, e.g., the E.164 format. This does not only ease the definition of subsequent conditions or maps, but it also converts the call to the format as required by the receiver.

The dialplan engine supports the following features:

• Number expansion and reduction: called, calling, and redirected numbers are converted to one of the following formats: international, national, local, or internal (extension-only) format; for each format, either prefix digits or digital number type flags may be used.
• Adding and removing of the line access code: If not present, dialed numbers are automatically prepended by the digit(s) needed to get access to the public telephone network.
• Support for North American numbering plan: Up to 10 area codes may be configured to be treated differently. For example, in many areas dialing into neighbouring areas requires to not dial a long-distance prefix.

Important information about the outside access digit configuration

• Configure the outside access digit only if there is a PBX between the PSTN and the Dialogic^® Diva^® SIPcontrol^TM Software, and if this PBX requires the outside access digit for external calls. If you need to configure the outside access digit, also configure the following related options:
  • Incoming PSTN access code provided by the PBX: This option defines whether the Diva SIPcontrol software expects the outside access digit in the calling number of external calls from the PBX.
    The PBX normally prepends the outside access digit to the calling number in incoming external calls in order to enable callback functionality at internal phones. If this is the case, enable this option.
  • PSTN access code provided by the SIP caller: This option defines whether the Diva SIPcontrol software expects the outside access digit in the called number of external calls from SIP to the PSTN.
    It is normally required to prepend the outside access digit to call an external number from an internal phone. In this case, these are phones on the SIP side. However, in some configurations this is not required, especially in a configuration that is part of the North American numbering plan (NANP), where an internal number can be identified based on its length. If it is possible to identify an internal call purely by the length of the called number, this option can be disabled. In all other configurations with outside access digits this option has to be enabled.
    It is recommended to have this option enabled in dialplans with the outside access digit set.
• The Diva SIPcontrol software's number normalization function does not remove outside access digits as a PBX can for external calls. If the Diva SIPcontrol software needs to behave like a PBX with an outside access digit for external calls, use the Address Map functionality in combination with a Routing module.

To add a dialplan, click Add. To configure an existing dialplan, click Details. To open the online help for a specific parameter, click the parameter and a window with the help text will pop up. You may configure the parameters shown in the graphic and explained below:

Address Maps

In general, address maps should be used for cases that are not covered by the dialplan. Possible scenarios are:

• set the calling number to that of the central office on SIP-to-PSTN calls,
• change the called extension to another value if an employee left,
• remove trunk prefixes while routing to a global voicemail server.

Each address map consists of a number of rules that are checked and applied from first to last until a matching rule is found that has the Stop on match option enabled. A rule matches only if all three expressions of that rule match. The order of the address maps is not important, but the order of the rules within a map is significant and can therefore be changed with the "arrow down" and "arrow up" buttons.

To add an address mapping configuration, click the Add button. To configure an existing address map, click the Details button. To open the online help for a specific parameter, click the parameter and a window with the help text will pop up.

You may configure the parameters shown in the graphic and explained below:

If expressions should match from the beginning, prepend the caret symbol ("^") at the beginning of the expression, for example:

Number: 1234567

Expression: ^123

Format: 4567

Result: 45674567

Cause Code Maps

Depending on the type of SIP peer selected, different default mapping tables are used, to adapt the Dialogic^® Diva^® SIPcontrol^TM Software's responses to the values expected by that peer.

If the internal default mapping table provided by the Diva SIPcontrol software does not fulfill your needs, e.g., because your local PBX uses non-standard cause codes, you may configure your own cause code mapping table, which will be checked before the default table is. See Cause code mapping for the cause/response code mapping table. If you create your own cause code mapping table, make sure to select it in the SIP Peer Configuration under Enhanced.

To add a cause code, click Add. To change the settings, click Details on the right hand side. To open the online help for a specific parameter, click the parameter and a window with the help text will pop up.

You may configure the parameters shown in the graphic and explained below:

Codec Profiles

To configure the codec list, click the Add button. To change the settings, click the Details button on the right hand side. If you create a codec profile, make sure to select it in the SIP Peer Configuration under Enhanced.

To open the online help for a specific parameter, click the parameter and a window with the help text will pop up.

You may configure the parameters shown in the graphic and explained below:

Registrations

SIP devices can communicate directly if the URL of both devices is known, but in general, SIP gateways are used in a network to enable functionalities such as routing, registration, authentication, and authorization.

Registration at a registrar server can be useful because in many cases, only the SIP address of a user is known but the location (SIP address of the device) is unknown or may change. A registrar server keeps track of the location of user agents from which the registrar server has received REGISTER requests. Thus, only the SIP address of the user needs to be sent to the registrar server, which then returns one or more contact addresses of the user.

If the Dialogic^® Diva^® SIPcontrol^TM Software is configured to use a registrar server, it registers with the server as soon as it is active. Thus, all local addresses configured for registration are registered with the server. You may use either a private registrar service or a public registrar server.

To configure a registrar server, click the Add button. To change the settings, click the Details button on the right hand side. To open the online help for a specific parameter, click the parameter and a window with the help text will pop up.

You may configure the parameters shown in the graphic and explained below:

To configure the settings for each user that should register at the same registrar server, click Add and configure the following parameters:

System Settings

To open the online help for a specific parameter, click the parameter and a window with the help text will pop up. You may configure the parameters shown in the graphic and explained below:

Data Security Overview

Since version 2.0, the Dialogic^® Diva^® SIPcontrol^TM software provides additional security options for transmitted and received data:

• Secure HTTP: You may use Secure HTTP (HTTPS) to transmit data between the web-based configuration interface of the Diva SIPcontrol software and your web browser.
• TLS: The Transport Layer Security (TLS) protocol may be used to encrypt and authorize SIP messages.
• Secure RTP: The Secure Real-time Transport Protocol (SRTP) may be used for encrypting the data of the actual conversation.

Note: The TLS protocols require digital identity certificates (e.g., public key certificates). See below for more information about certificates.

Secure HTTP

HTTP is a protocol that transmits data between the web-based configuration interface of the Diva SIPcontrol software and your web browser. Even though the HTTP interface has access security (via a password), the transmitted data is not entirely secure. The data is transmitted as clear text and thus it is possible for the transmission to be intercepted and, in turn, for the data to be read.

HTTPS, however, uses HTTP over an encrypted Secure Sockets Layer (SSL) or Transport Layer Security (TLS) connection and with a different default port than HTTP.

As an example, if a message containing a request to change a password was captured by a third party, the third party could log on to the Diva SIPcontrol software web interface and change the configuration. HTTPS encrypts and authenticates HTTP data, and thus the data is no longer transmitted as clear text and is not easily readable.

HTTPS requires two actions by the user:

• Both the Diva SIPcontrol software and the computer on which the web browser used to connect to the Diva SIPcontrol software via HTTPS is running must be configured with the proper certificate.
• When accessing the Diva SIPcontrol software web interface, use https:// instead of the non-secure http:// followed by the URL of the PC on which the Diva SIPcontrol software is installed.

TLS

SIP (Session Initiation Protocol) is a signaling protocol used for VoIP calls over the Internet. SIP messages contain information such as call-party information, call media type, whether it is a secure call, and if so, what encryption algorithm is used, etc. SIP can be carried by UDP, TCP, or TLS transports. Both UDP and TCP transport data in clear text. As a result, UDP and TCP can easily be monitored by a third party. TLS, on the other hand, carries SIP data in a secure way by encrypting the data and authenticating the transport connections. Authentication provides that you are talking to the intended peer. For authentication purposes, you need to install certificates as described in Security Profiles and enable TLS as transport protocol as described in Network Interfaces. For general information about certificates, see the section below.

Secure RTP

Once a Voice over IP (VoIP) call is established, voice data is transported in packets with the Real-time Transport Protocol (RTP). The voice data can be easily extracted from RTP packets and replayed using commercially available software. SRTP adds security by encrypting voice data and authenticating packets. Digital identity certificates are not required; the parameters are negotiated during call initiation time. SRTP mode is activated typically in combination with TLS, but in some cases (e.g., testing, intranet connections only) it is useful to allow SRTP also without TLS being activated.

For encryption and decryption of data, SRTP uses "so called" ciphers. The two parties involved in a conversation must be "compatible" in the sense that each party understands the other party's cipher requirements and supports them. The Diva SIPcontrol software supports the following ciphers: DH, ADH, AES (128-256 bits), 3DES (64 bits), DES (64 bits), RC4 (64bytes), RC4 (256 bytes), MD5, SHA1.

SRTP can be set for each SIP peer in the Security configuration. The cipher level can be set in the Global Security Parameters as described in Security Profiles.

Certificates

For authentication and data encryption, certificates need to be installed on the computer with the Diva SIPcontrol software and on remote computers. When a secure domain is opened, server and client authentify each other with a so called "SSL handshake". With this handshake, the identity of a user is certified and the user can be trusted. All necessary certificates are provided by a Certificate Authority (CA) and they are issued for one domain name. For test purposes or internal usage, you can also create and sign your own self-signed certificate, e.g., with one of the many tools available on the internet, just search for "self-signed certificate" and you will find a list of possible tools. But you need to be aware that self-signed certificates do not provide the same security as CA-signed certificates. All files need to be in "pem" format, that means base64 encoded. Also, many web browsers check if the certificate is signed by a CA, and, if it is not, a warning message will pop up asking whether the user really wants to trust that web site, which can make the user feel insecure.

A default certificate is provided with the software, but for security reasons, you should install your own web server certificate.

Certificate files can be generated in different formats, e.g., .pem, .der, .cer, or .pfx. All files need to be in "pem" format, that means base64 encoded, in order to be used by the Diva SIPcontrol software.

Note for CER files: CER files can be renamed to .pem directly if they are base64 encoded. No bag attribute lines and/or additional CR and empty lines are allowed. If CER files are ASN.1 coded, they need to be converted to with a converter tool.

Note for PFX files: The PFX or PKCS#12 format is a binary format for storing the server certificate, any intermediate certificates, and the private key in one encryptable file. When converting a PFX file to PEM format, tools like OpenSSL will put all the certificates and the private key into a single file. You will need to open the file in a text editor and copy each certificate and private key (including the BEGIN/END statments) to its own individual text file and save them as certificate.cer, CACert.cer, and privateKey.key respectively.

How to retrieve keys and certificates from a PFX file for use in the Diva SIPcontrol software

In the following procedure openssl is used as example converter tool.

1. Export the private key file from the PFX file:

   openssl pkcs12 -in filename.pfx -nocerts -out protected-key.pem

2. Remove the passphrase from the private key as required by the Diva SIPcontrol software:

   openssl rsa -in protected-key.pem -out key.pem

3. Export the certificate file from the PFX file:

   openssl pkcs12 -in filename.pfx -clcerts -nokeys -out cert.cer

4. Export the Root CA certificate file from the PFX file:

   openssl pkcs12 -in filename.pfx -cacerts -nokeys -out cacert.cer

How Calls Are Processed

The Dialogic^® Diva^® SIPcontrol^TM Software uses an endpoint-based approach to process calls, which means that every PSTN interface and every configured SIP peer is considered as a single endpoint. The endpoint saves the Diva SIPcontrol software settings for the respective PSTN interface or SIP peer. Each call originates at a specific endpoint (on the SIP side after assigning the SIP call request to one of the configured peers) and needs a route to find its designated endpoint (the destination). Thus, the most simple configuration needs one PSTN endpoint, one SIP peer, and one route as shown in red in the graphic below.

This graphic shows that an endpoint is only a virtual object of a real device. The endpoint saves the settings for the corresponding device. For example, if a call should be routed from SIP device 3 to PSTN device 2 as marked red in the graphic, then:

• The settings of SIP device 3 need to be configured as SIP peer endpoint in the SIP Peer Configuration,
• the settings PSTN device 2 needs to be configured as PSTN endpoint in the PSTN Interface Configuration, and
• the condition "called address is 3456" needs to be configured in the Routing Configuration to route the call to the correct device.

If you have for example a SIP or PSTN device 4 with no endpoints configured in the Diva SIPcontrol software, then you cannot establish a call, because the Diva SIPcontrol software will not know the settings of the device.

A PSTN endpoint is found via its controller number. On the SIP side, multiple SIP peers may connect via the same network interface. Therefore, the assignment is more complex:

1. The host/domain name and port number of the received "FROM" header is compared against the SIP peer settings.
2. If no host matches, the same address is compared against the "Domain" parameters of the SIP peers.
3. If no match is found, the Diva SIPcontrol software looks for a SIP peer with the Default SIP to PSTN Peer option enabled.
4. If the call cannot be assigned, regardless of whether the call originated in the PSTN or SIP network, the call is rejected.

Every route defines only one direction. Therefore, at least two routes are needed to support both PSTN-to-SIP and SIP-to-PSTN connections. The basic call (without address manipulation) is processed as follows:

Some things to know about call processing

• Each route may point to several destinations, between which the Diva SIPcontrol software chooses according to the following settings (in decreasing order of importance):
  • availability (destination enabled),
  • alive state of destination (if enabled to be verified),
  • priority (Master/Slave),
  • channel load quota (a factor calculated by comparing used vs. total supported channels).
• For each call only one route is chosen. Even if another route would also match the call criteria, only the first matching route is ever evaluated, so be careful when placing default routes at the beginning. Load-balancing/failover is only performed between the destinations of a single route.
• Routes without any conditions always match (as long as the source endpoint is listed in route sources).

Emergency calls

In many environments, certain numbers, e.g., 110/112 in Germany or 911 in the USA, have to be handled differently from others. For example, they might need to be dialed without any access digit.

This can be achieved by creating an additional route from any configured SIP peers to one or more PSTN interfaces and setting the called address expression to the emergency number(s). The route should be placed at the top position in the list. Should there be a dialplan and/or address map configured for the respective PSTN interfaces, it may be necessary to add another regular expression to the address maps of the interfaces to handle those calls.

Routing conditions

The Dialogic^® Diva^® SIPcontrol^TM Software organizes the conditions of a route in a list. Each list entry consists of different expressions for called, calling, and redirected address. The route matches only if all three expressions simultaneously match the respective call addresses. Empty expressions are considered to match, so there is no need to add wildcards into unused expressions. As a result, if a call should match either a called address or a calling number, two list entries have to be created, with called expression in the first and calling expression in the second row. If both have to match concurrently, both expressions have to be entered into the same list entry.

Routing examples

This section describes the configuration of four possible routing scenarios:

• Direct routing between one PSTN interface and one SIP peer
• Connect two SIP peers to two PSTN interface exclusively
• Connect two SIP peers to the same PSTN interface
• Load-balancing or failover between two SIP peers

Direct routing between one PSTN interface and one SIP peer

If you choose to route all calls from PSTN to the same SIP peer, and calls from that SIP peer to PSTN, configure the parameters as follows. For this configuration, no address rewriting is needed:

1. Under PSTN Interface Configuration, enable and configure all PSTN interfaces connected to a PBX. Confirm each dialog box with OK.
2. Under SIP Peer Configuration, create a SIP peer with the necessary settings and make sure that the option Default SIP to PSTN peer is enabled. Confirm with OK.
3. Under Routing Configuration, create route no. 1 and do the following:
   • Select PSTN to SIP as direction.
   • Enable all required PSTN interfaces.
   • Select the SIP peer configured in step 2 as the Master destination.
   • Set the parameter Number format to Unchanged.
   • Confirm with OK.
4. Under Routing Configuration, create route no. 2 and do the following:
   • Select SIP to PSTN as direction.
   • Enable the SIP peer configured in step 2 as source peer.
   • Enable all required PSTN interfaces as the Master destination.
   • Set the parameter Number format to Unchanged.
   • Confirm with OK.
5. Save the configuration in the main configuration interface.

Connect two SIP peers to two PSTN interfaces exclusively

If you choose to connect two SIP peers to two PSTN interfaces, so that each SIP peer may use one interface exclusively, then carry out the following configuration steps. The procedure is similar if you need to configure more PSTN interfaces, e.g., three PSTN interfaces to three SIP peers.

1. Under PSTN Interface Configuration, enable and configure the two PSTN interfaces. Confirm with OK.
2. Under SIP Peer Configuration, create both SIP peers and make sure the entry in Domain matches exactly the domain used by the SIP peer in its SIP address for outgoing calls. Do not enable the option Default PSTN to SIP Peer for any of these peers. Confirm with OK.
3. Under Routing Configuration, create route no. 1 and do the following:
   • Select PSTN to SIP as direction.
   • Enable the first PSTN interface as source
   • Enable the first SIP peer configured in step 2 as the Master destination.
   • Confirm with OK.
4. Under Routing Configuration, create route no. 2 and repeat step 3 for the second PSTN interface and the second SIP peer.
5. Under Routing Configuration, create route no. 3 and do the following:
   • Select SIP to PSTN as direction.
   • Enable the first SIP peer configured in step 2 as source peer.
   • Enable the first PSTN interfaces as the Master destination.
   • Confirm with OK.
6. Under Routing Configuration, create route no. 4 and repeat step 5 for the second PSTN interface and the second SIP peer.
7. Save the configuration in the main configuration interface.

Connect two SIP peers to the same PSTN interface

You want to connect two SIP peers to the same PSTN interface so that all calls from the PSTN are sent to the first SIP peer if the numbers begin with "1" and to the second peer if the numbers begin with "2".

1. Under PSTN Interface Configuration, enable and configure the PSTN interface. Confirm with OK.
2. Under SIP Peer Configuration, create both SIP peers and make sure the entries in Domain matches exactly the domain used by the SIP peer in its SIP address for outgoing calls. Do not enable the option Default PSTN to SIP Peer for any of these peers. Confirm with OK.
3. Under Routing Configuration, create route no. 1 and do the following:
   • Select PSTN to SIP as direction.
   • Enable the first PSTN interface as source.
   • Enable the first SIP peer configured in step 2 as the Master destination.
   • Under Conditions click Add and set the Called address to "1.*".
   • Confirm with OK.
4. Under Routing Configuration, create route no. 2 and repeat step 3 for the second SIP peer with the only difference that the called address condition for this route is "2.*".
5. Under Routing Configuration, create route no. 3 and do the following:
   • Select SIP to PSTN as direction.
   • Enable both SIP peers as source peer.
   • Enable the first PSTN interfaces as the Master destination.
   • Confirm with OK.
6. Save the configuration in the main configuration interface.

If calls other than those beginning with 1 or 2 should also be directed to one peer, remove the condition from the respective PSTN to SIP route and move the route to the end of the list.

Load balancing or failover between two SIP peers

If two SIP servers should be configured as load-balancing or failover, configure the following:

1. Under PSTN Interface Configuration, enable and configure all required PSTN interfaces. Confirm with OK.
2. Under SIP Peer Configuration, create both SIP peers and make sure the entries in Domain matches exactly the domain used by the SIP peer in its SIP address for outgoing calls. Do not enable the option Default PSTN to SIP Peer for any of these peers. If you configure a failover, SIP peer 1 (the master) should have the option Alive check enabled. Confirm with OK.
3. Under Routing Configuration, create route no. 1 and do the following:
   • Select PSTN to SIP as direction.
   • Enable the first PSTN interface as source.
   • Enable the first SIP peer configured in step 2 as the Master destination. For load-balancing configurations, SIP peer no. 2 should be configured as the Master destination and for failover configurations it should be configured as Slave destination.
   • Confirm with OK.
4. Under Routing Configuration, create route no. 2 and do the following:
   • Select SIP to PSTN as direction.
   • Enable both SIP peers as source peer.
   • Enable the first PSTN interfaces as the Master destination.
   • Confirm with OK.
5. Save the configuration in the main configuration interface.

If calls other than those beginning with 1 or 2 should also be directed to one peer, remove the condition from the respective PSTN to SIP route and move the route to the end of the list.

How Call Addresses Are Processed

The call addresses provided by the caller may be modified at different stages of the call processing within the Dialogic^® Diva^® SIPcontrol^TM Software. The reason for multiple manipulation is that it allows for modifying the address where it is needed, which means that more complex environments can be configured with less effort, since data does not need to be entered redundantly at different places. It also makes it easier to "team" SIP peers or PSTN interfaces with different settings.

The Diva SIPcontrol software converts addresses automatically without any intervention from the user. This means, that the Diva SIPcontrol software adds or removes a special prefix to a number with a known number type, e.g. "+" for international numbers, when converting between a number and an address. See Number modification using regular expressions for a list of common formats.

Note: Number type flags from digital networks, e.g., ISDN or SS7 are converted into special prefixes on the SIP side. International numbers get a "+" prefix, national numbers get an "N" prefix, and subscriber numbers get an "S" prefix.

The automatic conversions are done for calling numbers, called numbers, and redirected numbers.

Possible scenarios:

• At a PSTN interface, a line access digit must be prepended in order to call to the public network, while another PSTN interface is directly connected and does not need an access digit.

  Solution: Add a regular expression to outbound address map of the first interface.

• All calls to a number beginning with "9" shall be routed to one specific SIP peer while removing this digit.

  Solution: Manipulate the called number in the route. This way the SIP peer may also receive calls to other numbers (via other routes) without having to deal with different number formats.

• SIP peer "A" needs the dialed numbers to be formatted in E.164 format, while SIP peer "B", which is in load-balancing or fail-over partnership with "A", needs it in an extension-only format.

  Solution: Define different number formats in the SIP peer settings.

• SIP peer "A" is located at a different location than SIP peer "B", e.g., London and Stuttgart; therefore, both need different location settings regarding country and area codes, etc.

  Solution: Create different dialplans and assign each dialplan to one SIP peer.

Address manipulation is done as follows

Note: Each step is optional.

1. Save the inbound call addresses as "A".
2. Apply the "address map inbound" of the endpoint assigned to the call setup request to "A", resulting in "B".
3. To check the first route: apply the number format settings of the route together with the dialplan of the source endpoint to the call addresses "B", resulting in "C".
4. Check the route as described in the route processing section (5) against addresses "C". If the route does not match, discard the changes and try the next route with "B" again.
5. If the route matches, apply the route address map to the addresses "C", resulting in "D".
6. After selecting one of the destinations of the route, normalize the addresses "D" using the dialplan and number format of the destination endpoint, resulting in addresses "E".
7. Apply the outbound address map of the destination endpoint to "E", giving the effective call addresses "F" sent to the destination.
8. If the call to the selected destination endpoint fails and if there are other endpoints in a fail-over configuration, start with step 6 again with the respective settings of the next endpoint.

How Address Maps Are Processed

Address maps are processed as follows:

1. Get the first map rule of the address map.
2. Verify if called, calling, and redirect expression each match the respective part of the call addresses (or are empty). If not, verify the next map rule.
3. If all three expressions match, apply each format string of the rule to the respective address match.
4. If the option Stop on match is enabled, stop processing. Otherwise, continue with the next rule as described in step 2.

How Numbers Are Processed

The Dialogic^® Diva^® SIPcontrol^TM Software provides two mechanisms for number processing. Both mechanisms can be used together:

1. Number normalization based on a Dialplan
2. Number modification using regular expressions

Number normalization based on a Dialplan

The number normalization based on a dialplan can work in an environment in which the Dialogic^® Diva^® SIPcontrol^TM Software is connected to a private SIP network and a public switched telephone network (PSTN), optionally with a PBX between the PSTN and the Diva SIPcontrol software. If the Diva SIPcontrol software is used as a gateway between a private circuit switched network and a public SIP-based network, the number normalization function of the Diva SIPcontrol software should not be used.

The Diva SIPcontrol software also supports dialplans using the North American numbering plan (NANP). See Dialplan Configuration for more information.

The number normalization is done in two steps:

1. The received called, calling and redirected numbers are analyzed based on the dialplan configured for the PSTN Interface or SIP Peer.
2. The number is converted into the configured target format. Six target formats are available:
   • International number with prefixes: All numbers are converted to an international number with the prefix for international calls and, if required, an outside access digit.
   • International number with number type: All numbers are converted to an E.164 number with the number type flag set to "international" ("+" is used in SIP addresses).
   • National number with prefixes: If possible, all numbers are converted to a national number with the prefix for national calls and an outside access digit, as required. Exception: Numbers with different country code will be converted to an international number with prefix for international calls and outside access digit, as required.
   • National number with number type: If possible, all numbers are converted to a national number with the number type flag set to "national". Exception: Numbers with different country code will be converted to an international number with number type set to "international".
     Note: This target format should not be used for calls to SIP networks.
   • Extension only with prefixes: All numbers are reduced as much as possible; only the required prefixes are prepended.
   • Extension only with number type: All numbers are reduced as much as possible. Instead of prefixes the appropriate number type is set.
     Note: This target format should not be used for calls to SIP networks.

Important information about the outside access digit configuration

• Configure the outside access digit only if there is a PBX between the PSTN and the Dialogic^® Diva^® SIPcontrol^TM Software, and if this PBX requires the outside access digit for external calls. If you need to configure the outside access digit, also configure the following related options:
  • Incoming PSTN access code provided by the PBX: This option defines whether the Diva SIPcontrol software expects the outside access digit in the calling number of external calls from the PBX.
    The PBX normally prepends the outside access digit to the calling number in incoming external calls in order to enable callback functionality at internal phones. If this is the case, enable this option.
  • PSTN access code provided by the SIP caller: This option defines whether the Diva SIPcontrol software expects the outside access digit in the called number of external calls from SIP to the PSTN.
    It is normally required to prepend the outside access digit to call an external number from an internal phone. In this case, these are phones on the SIP side. However, in some configurations this is not required, especially in a configuration that is part of the North American numbering plan (NANP), where an internal number can be identified based on its length. If it is possible to identify an internal call purely by the length of the called number, this option can be disabled. In all other configurations with outside access digits this option has to be enabled.
    It is recommended to have this option enabled in dialplans with the outside access digit set.
• The Diva SIPcontrol software's number normalization function does not remove outside access digits as a PBX can for external calls. If the Diva SIPcontrol software needs to behave like a PBX with an outside access digit for external calls, use the Address Map functionality in combination with a Routing module.

Number modification using regular expressions

The Dialogic^® Diva^® SIPcontrol^TM Software organizes regular expressions into address maps, and each endpoint or route may be assigned one map. Each address map contains a number of regular expressions together with the respective output format string that ensures that virtually every required manipulation scheme can be configured.

By using separate address maps, instead of rules embedded into the routes and endpoints, it is possible to share the same settings across different objects. For example, if several PSTN interfaces are connected to the same PBX, they will most probably be configured with the same settings and, therefore, can share an address map that the Diva SIPcontrol software lets you assign for each individual controller.

The Diva SIPcontrol software uses the style of regular expressions used by Perl. Most tutorials and how-to's covering Perl regular expressions can apply to the Diva SIPcontrol software.

Common expressions:

Common formats:

Examples

Note: In all examples, the hyphen ("-") is only used for clarification. It must not be included either in the dialed numbers or in the configured expressions and formats.

The examples may be used for calling or called number normalization for both the inbound and outbound directions.

Omit the prefix digits

The leading prefix "33" should be removed from the number 33-444-5555 and thus be converted into 444-5555.

Note: If the number does not start with "33", it passes unchanged.

Expression entry: ^33

Format entry: (none)

Add the prefix digits

The number 444-5555 needs the leading prefix "9" and should be dialed as 9-444-5555.

Expression entry: .*

Format entry: 9$&

Replace an international number type by prefix

A call indicated as an international call should be placed with prefixes instead.

Example entry: The number +1-472-333-7777 should be dialed as 011-472-333-7777

Expression entry: ^\+

Format: 01

Replace the international dial prefix by number type

A call with an international dial prefix should be placed with an international number type instead.

Example: The number (01)1-472-333-7777 should be dialed as +1-472-333-7777

Expression entry: ^01

Format entry: +

Replace an extension by another

Calls for specific extensions should be indicated with other extensions, e.g., the extension 1111 should be replaced by 2222, and extension 3333 by extension 4444.

Note: In calls from PSTN to SIP, the dialed SIP peer remains the same although the number is replaced.

First expression entry: 1111(@.*)?$

First format entry: 2222

Stop on Match: true

Second expression entry: 3333(@.*)?$

Second format entry: 4444

Stop on Match: true

Note: This example applies only on calls from SIP to the PSTN.

Replace the "N" in a national number

The "N" can be set to signal a number as national number.

Task: Replace the "N" in a national number with the national prefix.

Example: N123-45678 should be signaled as 0123-45678

Expression: ^N

Format entry: 0

Display the "user=phone" parameter without E.164

The "user=phone" parameter is set automatically if the number is a valid "tel:" URI. The number is either in E.164 format or has the "phone-context=XXX" parameter added. If you need the "user=phone" without E.164, you need to provide the phone-context parameter.

Task: Display "user=phone" parameter without E.164 and provide phone-context parameter.

Example: Present the phone number +1(123)727-0203 without E.164.

Expression: ^(.*)

Format entry: $1;phone-context=+1(123)$1

Software Uninstallation

The Dialogic^® Diva^® SIPcontrol^TM software is uninstalled automatically when the Dialogic^® Diva^® for Linux software package is uninstalled.

Cause Code Mapping

The Dialogic^® Diva^® SIPcontrol^TM Software includes a default cause/response code mapping table that includes the most common cause codes according to RFC 3398 and RFC 4497. If you need to define a cause code mapping other than in the table, you can configure it in the Cause Code Configuration.

For ISDN to SIP code mappings, see ISDN cause code to SIP response code.

For SIP to ISDN code mappings, see SIP response code to ISDN cause code.

ISDN cause code to SIP response code

SIP response code to ISDN cause code

Event Logging

A computer with the Dialogic^® Diva^® SIPcontrol^TM software installed, may write the following types of events into the System Event Log:

• Errors
• Warnings
• Informational Messages

Errors

An error is a significant problem such as loss of data or loss of functionality. For example, if a service fails to load, an error event will be logged.

See the following table for possible error events. Variables are enclosed in angle brackets. Parameters enclosed in square brackets are optional:

Warnings

A warning is an event that is not necessarily significant. But it might indicate a possible future problem.

See the following table for possible warnings. Variables are enclosed in angle brackets:

Informational messages

Informational messages refer to successful operation events such as starting or stopping the service:

See the following for informal events. Variables are enclosed in angle brackets:

Use Case for Dialogic® HMP software

This use case describes the usage of the Dialogic^® Host Media Processing (HMP) software running on the same computer as the Diva Media Board and the Diva SIPcontrol software as shown in the graphic below. However, the Diva SIPcontrol software supports also the interoperability with HMP over the LAN. The use case is based on Diva SIPcontrol software version 1.8 and on HMP version 3.0WIN and 3.1LIN. In order for the application based on the Dialogic^® Global Call API to connect with the Diva SIPcontrol software, it needs to be set to listen on port 5060 and to send SIP messages to the IP address 127.0.0.1 on port 9803.

To configure the Diva SIPcontrol software to function with your Global Call application:

1. Open the Diva SIPcontrol software web interface to configure the required settings. To do so, click Start > Programs > Dialogic Diva > SIPcontrol Configuration.
2. In the Diva SIPcontrol web interface, click SIPcontrol configuration on the left hand side to open the SIPcontrol Configuration page. For this configuration scenario, the network interface, a SIP peer, and two routings need to be configured.
3. Under Network Interface Configuration, set the Local Loopback Interface to Enabled, and as SIP Listen Port enter 9803.
4. Open the SIP Peer Configuration, click Add, and configure the following parameters:

   

   Under Edit SIP Peer Configuration, configure the following parameters:

   • Name: Enter a unique name to easily identify the SIP peer.
   • Peer type: Leave at Default setting.
   • Protocol: Select UDP.

     Under Enhanced Configuration, enable the option Default SIP to PSTN peer.

     Click OK to save the settings and to close the window.

5. Create two routings, one for each direction (SIP to PSTN and PSTN to SIP). To configure the routing from SIP to PSTN, open the Routing Configuration, click Add, and configure the following parameters:

   

   • Name: Enter a unique name to easily identify the routing.
   • Direction: Select SIP to PSTN from the drop down menu.
   • Select sources: Select the above configured SIP peer as source.
   • Select destinations: Select the controllers of the Dialogic^® Diva^® Media Board.

     Click OK to save the settings and to close the window.

6. For the routing from PSTN to SIP, click Add again, and configure the following parameters:

   

   • Name: Enter a unique name to easily identify the routing.
   • Direction: Select PSTN to SIP from the drop down menu.
   • Select sources: Select the controllers of the Diva Media Board.
   • Select destinations: Select the above configured SIP peer as master destination.

     Click OK to save the settings and to close the window.

7. Click Save in the main configuration page to save the settings and to activate the changes.