Author: VoIP Info
Sipura Documentation Line 1
Posted: October 15, 2005 Sipura Line 1 Webpage Documentation Line Enable Streaming Audio Server (SAS) SAS Enable SAS DLG Refresh Intvl SAS Inbound RTP Sink NAT Settings NAT Mapping Enable : Informs the SPA that it should discover the public…
Read More »Sipura Documentation Regional
Posted: October 15, 2005 Sipura Regional Webpage Documentation Call Progress Tones Dial Tone Second Dial Tone Outside Dial Tone Prompt Tone Busy Tone Reorder Tone Off Hook Warning Tone Ring Back Tone Confirm Tone SIT1 Tone SIT2 Tone SIT3 Tone SIT4 Tone…
Read More »Sipura Documentation Provisioning
Posted: October 15, 2005 Sipura Provisioning Webpage Documentation Configuration Profile Provision Enable Resync On Reset Resync Random Delay Resync Periodic Resync Error Retry Delay Forced Resync Delay Resync From SIP Resync After Upgrade Attempt Resync Trigger 1 Resync Trigger 2 Resync…
Read More »Sipura Documentation SIP
Posted: October 15, 2005 Sipura SIP Webpage Documentation SIP Parameters Max Forward Max Redirection Max Auth SIP User Agent Name SIP Server Name SIP Accept Language DTMF Relay MIME Type Hook Flash MIME Type Remove Last Reg Use Compact Header …
Read More »Sipura Documentation Info
Posted: October 15, 2005 Sipura Info Webpage Documentation System Information DHCP Current IP Host Name Domain Current Netmask Current Gateway Primary DNS Secondary DNS Product Information Product Name Serial Number Software Version Hardware Version MAC Address Client Certificate Customization System Status Current…
Read More »Sipura Documentation
Posted: October 15, 2005 Page Contents Purpose Layout of the Documentation Example Configurations Advanced/admin Web page Tabs Other notes about configuration Disable tone when another incoming call arrives Purpose This page tries to better document the configuration parameters of the Sipura SPA…
Read More »SER example outboundproxy
Posted: October 14, 2005 SIP Express Router example: SER as a multihomed outbound proxy This is a stripped-down version of the ser.cfg file that powers a SIP application-level gateway (ALG). This configuration is based on the OnSIP Getting Started 5.0 configuration and…
Read More »Asterisk IAXmodem
Posted: October 14, 2005 IAXmodem Main purpose: Connect Hylafax to IAXmodem. https://sourceforge.net/projects/iaxmodem IAXmodem is a software modem written in C that uses an IAX channel (commonly provided by an Asterisk PBX system) instead of a traditional phone line and uses a DSP…
Read More »D-Link DPH-140S
Posted: October 13, 2005 Business IP Phone (SIP) Product Features: “¢ Supports STUN and Outbound Proxy “¢ Works both on public IP or behind NAT “¢ Make VoIP Phone Calls over the Internet and Save on Long Distance Charges* “¢ Speakerphone for…
Read More »say datetime
Posted: October 13, 2005 SAY DATETIME <time> <escape digits> [format] [timezone] Say a given date and time, returning early if any of the given DTMF digits are received on the channel. <time> is number of seconds elapsed since 00:00:00 on January 1,…
Read More »say date
Posted: October 13, 2005 SAY DATE <date> <escape digits> Say a given date, returning early if any of the given DTMF digits are received on the channel. <date> is number of seconds elapsed since 00:00:00 on January 1, 1970, Coordinated Universal Time…
Read More »IAX2 ping tool
Posted: October 13, 2005 !/usr/bin/perl -w udp ping tool use IO::Socket;$target = shift; #”192.168.0.255″;$target_port = 4569; socket(PING, PF_INET, SOCK_DGRAM, getprotobyname(“udp”)); Build Packet … Names from ethereal filter of registration packet $src_call = “8000”; #8000 most siginificant bit is IAX packet type full…
Read More »VOIP for business
Posted: October 11, 2005 VoIP for business 22 – 23 March 2006 Olympia Conference Centre, Level 3, London If you are involved in shaping your company’s strategic communications needs around mobilising the enterprise, increasing productivity and ensuring business growth and continuity, then…
Read More »Asterisk cmd SipAddHeader
Posted: October 11, 2005 Add a SIP header to the outbound call Description SIPAddHeader(Header: Content) Adds a header to a SIP call placed with DIAL. Remember to user the X-header if you are adding non-standard SIP headers, like “X-Asterisk-Accountcode:”. Use this with…
Read More »ICS France Dialup IP-Phone
Posted: October 10, 2005 Top-Phone 631S (SIP) / Top-Phone 631H (H.323) Built-in dial-up modem Top-Phone 631 IP Phone with Ethernet and built-in modem interfaces Connects directly to the internet via a modem, hub, or router Outstanding voice quality that is very comparable…
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