5/16/2023 0 Comments Simplefax device siptrunk.comĥ Appendix C: DIGIUM END-USER PURCHASE AND LICENSE AGREEMENT.55 Digium, Inc. Ĥ Chapter 4: Troubleshooting Manager Events Fax Transmission Completion Events Fax Status Events T.38 Fax Status Events Fax Document Status Events Manager Actions FaxLicenseList Action FaxLicenseStatus Action Asterisk Command Line Interface (CLI) fax set debug on fax set debug off fax set g711cap off fax set g711cap on fax set t38cap off fax set t38cap on fax show capabilities fax show hostid fax show licenses fax show session fax show sessions fax show settings fax show stats fax show version Frequently Asked Questions.33 Appendix A: Dialplan Examples.39 A.1 Simple Fax Transmit / Receive.39 A.2 Trunk, app_fax, and spandsp A.3 Asterisk 1.4, agx-ast-addons, and spandsp Appendix B: Glossary and Acronyms.44 Digium, Inc. ģ TABLE OF CONTENTS Chapter 1: Overview What is Asterisk? Asterisk as a Phone Switch (PBX) Asterisk as a Gateway Asterisk as a Feature/Media Server Asterisk in the Call Center Asterisk in the Network Asterisk Everywhere.8 Chapter 2: Installation Installation Overview Register Fax For Asterisk Determine Optimum Build Determine Required Components Install res_fax Install res_fax_digium Load Fax For Asterisk Modules Verify Installation Backup License File.16 Chapter 3: Configuration Application Interfaces FAXOPT Function SendFAX Application ReceiveFAX Application res_fax.conf res_fax_nf Compatibility with spandsp.24 Digium, Inc. Any other trademarks mentioned in the document are the property of their respective owners. Asterisk, Digium, Switchvox, and AsteriskNOW are registered trademarks and Asterisk Business Edition, AsteriskGUI, and Asterisk Appliance are trademarks of Digium, Inc. Adobe and Acrobat are registered trademarks, and Acrobat Reader is a trademark of Adobe Systems Incorporated. This document has been prepared for use by professional and properly trained personnel, and the customer assumes full responsibility when using it. The manufacturer s liability for any errors in the documents is limited to the correction of errors and the aforementioned advisory services. The manufacturer will, if necessary, explain issues which may not be covered by this documentation. has made every effort to ensure that the instructions contained in this document are adequate and error free. No part of this publication may be copied, distributed, transmitted, transcribed, stored in a retrieval system, or translated into any human or computer language without the prior written permission of Digium, Inc. Toll Free: Sales: Digium, Inc All rights reserved. 445 Jan Davis Drive NW Huntsville, AL United States Main Number: Tech Support: U.S. Your SIP device should only accept RTP traffic for a SIP call which is active, so the forwarding in tip 2, above, should notbe accompanied with blocking traffic from certain IP addresses (see here for an extended explanation).Ĥ.1 Fax For Asterisk Administrator Manual Rev. You should forward all RTP ports used by your device to the private IP address of your device if it is behind NATģ. You may choose to only allow incoming SIP traffic from the IP addresses listed in the "Gateways" section aboveĢ. Network security can be a consideration when implementing a VoIP solution. STATIC IP: your internet service provider (ISP) may provide a static IP at an additional cost STUN: if your PBX or softphone supports it, you may connect to our STUN server at on port 3478ĪLG: If your router or firewall is capable of properly implementing ALG, enabling it may alleviate your issue If it becomes apparent that you may be having a NAT issue, there are three possible solutions: Try to connect your PBX or softphone through our service without any special NAT configuration. Most customers are behind some form of NAT, and many do not have a static IP address. NAT stands for Network Address Translation. Since our SIP gateways are just a proxy, the audio can be delivered from various IP addresses and many different ports. which is located at 205.251.137.154īoth gateways will only accept SIP traffic on UDP port 5060Īudio related to SIP calls is delivered via RTP over UDP. SIP.TRUNK has two SIP Gateways which you may connect to: This is a set of basic networking requirements and terminologies for connecting to
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