Asterisk Voicemail Dialplan
Posted: May 16, 2004 How to transfer or forward to voicemail mailboxes Assume you have 3 digit extensions 100 – 199, then you can put something like this in the dialplan to allow users to transfer calls directly to a mailbox by…
Read More »Asterisk quotes
Posted: May 15, 2004 Various quotes from the Asterisk community From the CVSModified Files:chan_sip.cLog Message:The SIP motto is “There’s More Than One Standard for Doing It”markster John Todd:If Answer() clears variables, then this is a bug, where “bug” is defined as “behavior…
Read More »Asterisk config rtp.conf
Posted: May 15, 2004 rtp.conf Configuration of Asterisk Real Time Protocol, RTP, media channels. RTP is used for SIP communication. Details On your router you might want to arrange both traffic shaping (QoS) and port forwarding (in case of NAT) for the…
Read More »Telchemy
Posted: May 15, 2004 https://www.telchemy.com/ Telchemy is the developer of VQmon, a widely used VoIP call quality monitoring and analysis technology. VQmon/EP is integrated into IP phones, gateways and soft clients. VQmon/SA is integrated into probes, analyzers and routers. Telchemy also led…
Read More »Asterisk config enum.conf
Posted: May 14, 2004 This configuration file is used only by Asterisk cmd EnumLookup deprecated in Asterisk 1.4 See Asterisk func enumlookup for the replacement. Configuration of ENUM lookups See also EnumLookup Application Asterisk config files Example ; ENUM Configuration for resolving…
Read More »iKall LAN Phones
Posted: May 14, 2004 LinkTrans International Group Inc., a Canada based Internet-based telephony product and service provider, supplies global customers the advanced iKall? solutions based on its leading edge VoIP technology and customer-oriented service capability. iKall? customers can enjoy FREE domestic and…
Read More »Sipp – SIP Performance Tester
Posted: May 13, 2004 Sipp is a performance tester for the SIP protocol. It comes with a few basic SipStone user-agents scenarios (UAC & UAS), establishing and releasing multiple calls with the INVITE and BYE methods. Asterisk configuration for SIP (non-rtp listening)…
Read More »PeerCall
Posted: May 13, 2004 PeerCall might be as first identified as Skype hardware equivalent. It is telephony based on a P2P environment. Built upon Mirrador chip technology, it allows users to place free calls anywhere in the world to anybody equipped with…
Read More »Centile
Posted: May 12, 2004 https://www.centile.com From website: Founded in 1998, Centile Telecom Applications is the leading European developer of unified communications platforms for operators. The Centile Any3 architecture – Any service, over Any network, on Any terminal – addresses the needs of…
Read More »Asterisk cmd SubString
Posted: May 8, 2004 Note: This command is completely obsolete and has been removed Synopsis: Save substring digits in a given variable Description: SubString(variable=string_of_digits|count1|count2) Assigns the substring of string_of_digits to a given variable. Parameter count1 may be positive or negative. If it’s…
Read More »GL Communications
Posted: May 7, 2004 http://www.gl.com/voip Vendor of VOIP simulation and analysis systems Packet over IP and Ethernet Analysis and Simulation SIP RTP Bulk Call Generator All-IP Signaling and Traffic Monitoring Software Ethernet / VLAN / MPLS / IP / UDP Tester –…
Read More »CLID DTMF Sweden
Posted: May 6, 2004 DEFINITIONS Z interface is a term used in this document for all types of analogue 2-wire interfaces at the network operator side of analogue subscriber lines delivering the PSTN service. The Z interface can be located in local…
Read More »WorldGate
Posted: May 4, 2004 Manufacturer of the Ojo videophone From the website: The Ojo™ personal video phone leverages an enhanced version of the H.264 digital compression standard and high fidelity full duplex speakerphone technology. The 30 frames per second video can be…
Read More »Asterisk G.723 pass-thru
Posted: May 3, 2004 How to use the G.723.1 codec in pass-thru mode in Asterisk G.723.1 can only be used in pass-thru mode. This means that there can be no access to Asterisk cmd VoiceMail, etc In order to ensure that it…
Read More »Asterisk sip permit-deny-mask
Posted: April 28, 2004 IP address mask for SIP traffic In a user/peer definition in Asterisk config sip.conf, you may limit SIP traffic to and from this peer to a certain IP or network Syntax permit=<ipaddress>/<network mask> deny=<ipaddress>/<network mask> Order Matters! –…
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