VoIP Blog - Page 704

Note: These versions have reached End of Life. For the latest version click here. (“Taken from the README in libss7”) Tested Switches: ================ Siemens EWSD – (ITU style) MTP2 and MTP3 comes up, ISUP inbound and outbound calls work…

Note: These versions have reached End of Life. For the latest version click here. The Asterisk Speech Recognition API =================================== The generic speech recognition engine is implemented in the res_speech.so module. This module connects through the API to speech…

Note: These versions have reached End of Life. For the latest version click here. Asterisk SNMP Support ——————— Rudimentary support for SNMP access to Asterisk is available. To build this, one needs to have Net-SNMP development headers and libraries…

Note: These versions have reached End of Life. For the latest version click here. * The SMS application ——————— SMS() is an application to handles calls to/from text message capable phones and message centres using ETSI ES 201 912…

Note: These versions have reached End of Life. For the latest version click here. =============================================================================== =============================================================================== === Asterisk SMDI (Simple Message Desk Interface) integration ================= =============================================================================== =============================================================================== =============================================================================== ===== 1) Accessing SMDI information in the dialplan. ========================== =============================================================================== There…

Note: These versions have reached End of Life. For the latest version click here. Asterisk SIP/TLS Transport ========================== When using TLS the client will typically check the validity of the certificate chain. So that means you either need a…

Note: These versions have reached End of Life. For the latest version click here. What is the problem with SIP retransmits? —————————————– Sometimes you get messages in the console like these: – “retrans_pkt: Hanging up call XX77yy – no…

Note: These versions have reached End of Life. For the latest version click here. Overview ——- Asterisk currently supports configurable RTP packetization per codec for select RTP-based channels. Channels ——- These channel drivers allow RTP packetization on a user/peer/friend…

Note: These versions have reached End of Life. For the latest version click here. Real-time text in Asterisk ————————– The SIP channel has support for real-time text conversation calls in Asterisk (T.140). This is a way to perform text…

Note: These versions have reached End of Life. For the latest version click here. Asterisk Call Queues ——————– <template holder while we wait for input on a good README for call queues. Please open a bug report and add…

Note: These versions have reached End of Life. For the latest version click here. \begin{verbatim} DIGIUM GENERAL PEERING AGREEMENT (TM) Version 1.0.0, September 2004 Copyright (C) 2004 Digium, Inc. 445 Jan Davis Drive, Huntsville, AL 35806 USA Everyone…

Note: These versions have reached End of Life. For the latest version click here. OSP User Guide for Asterisk V1.6 9 February 2007 Table of Contents Revision History 1 Introduction 2 OSP Toolkit 2.1 Build OSP Toolkit 2.1.1 Unpacking…

Note: These versions have reached End of Life. For the latest version click here. All modules must have at least the following: static int load_module(): Do what you need to do when you get started. This function can return…

Note: These versions have reached End of Life. For the latest version click here. Changes to manager version 1.1: ——————————- * SYNTAX CLEANUPS —————– – Response: headers are now either “Success” – Action OK, this message contains response “Error”…