Author: VoIP Info
Asterisk FastAGI
Posted: June 20, 2005 FastAGI Implements the Asterisk Gateway Interface (AGI) over TCP sockets. This can help alleviate CPU load on your telephony server by relocating resource hungry scripts to another networked server. In order to instruct Asterisk to attempt a network…
Read More »Asterisk cmd MeetMe
Posted: June 20, 2005 MeetMe Synopsis MeetMe conference bridge Description MeetMe([confno][,[options][,pin]]): Enters the user into a specified MeetMe conference If the conference number is omitted, the user will be prompted to enter one. Page Contents Option details Option ‘s’ Option ‘q’ Option…
Read More »Mobicents
Posted: June 19, 2005 Mobicents is the First and Only Open Source JAIN SLEE 1.0 Certified product. Mobicents is a professional open source VoIP Middleware platform, which brings to telecom application developers what J2EE brings to Web and Enterprise application developers. In…
Read More »SFLphone
Posted: June 17, 2005 Overview SFLphone is a SIP/IAX2 compatible softphone for Linux. The SFLphone project’s goal is to create a robust enterprise-class desktop phone. While it can serve home users very well, it is designed with a hundred-calls-a-day receptionist in mind….
Read More »exec
Posted: June 17, 2005 Usage: EXEC <application> <options> Executes with given <options>. Applications are the functions you use to create a dial plan in extensions.conf. Returns:failure: 200 result=-2success: 200 result=<ret> <ret> is whatever the application returns EXAMPLE: EXEC Dial Zap/g1/123456 Also, you…
Read More »SER module domain
Posted: June 17, 2005 Domain module implements checks that based on domain table determine if a host part of an URI is “local” or not. A “local” domain is one that the proxy is responsible for. Domain module operates in caching or…
Read More »get variable
Posted: June 17, 2005 Usage: GET VARIABLE <variablename> Does not work* with global variables. Does not work with some variables that are generated by modules.(* now works for global variables in 1.2.10 – see http://bugs.digium.com/view.php?id=7609 ) ‘Variable’ actually includes functions (but no…
Read More »Asterisk FWD Assistant
Posted: June 17, 2005 FWD Assistant A setup assistant to add a Free World Dialup account to the configuration of an Asterisk server running on MacOSX. Limitations The assistant is only aware of phones and extensions created with the New Extension Assistant….
Read More »SIP Info DTMF
Posted: June 17, 2005 The SIP INFO method can be used by SIP network elements to transmit digits out-of-band as telephone-events in a reliable manner independent of the media stream. The SIP INFO method contains a message body in addition to the…
Read More »Asterisk phone grandstream budgetone
Posted: June 16, 2005 Tips for Grandstream Budgetone and Asterisk Use dtmfmode=rfc2833 in your sip.conf for your Grandstream BudgeTone and configure the GS to use “info” DTMF signalling. This provides a workaround to hangups due to problems in the GS (see Asterisk…
Read More »Asterisk cmd AgentLogin
Posted: June 15, 2005 Synopsis: Call agent login Description: AgentLogin([AgentNo][|options]) Asks the agent to login to the system. Always returns -1. While logged in, the agent can receive calls and will hear a ‘beep’ when a new call comes in. The agent…
Read More »PBX Automatic Call Delivery
Posted: June 15, 2005 Automatic Call Delivery ACD is a feature used to route calls in a call center environment to the appropriate agents, based on factors such as time available, skill sets, and priority levels. ACD systems place calls into a…
Read More »Asterisk config agents.conf
Posted: June 15, 2005 agents.conf ACD distributes incoming calls to the agents of a Queue. Agents are configured in the file queues.conf: New in Asterisk v1.2.0: The default for ackcall has been changed to “no” instead of “yes” because of a bug…
Read More »Asterisk Agents
Posted: June 15, 2005 Asterisk call agents Agents are human beings – well, phone extensions that are used by human beings. Setting up agents in the Asterisk config agents.conf file allows you to then assign agents in your call queues as a…
Read More »Asterisk Manager API Action QueueAdd
Posted: June 15, 2005 Action: QueueAdd Parameters: Queue: Existing queue to add member Interface: Member interface (sip/1000, zap/1-1, Local/1000@LocalNumber etc)? Penalty: Entries with higher penalties are considered last Paused: Boolean for member initially paused or not MemberName: Username (Mark Spenser) “asterisk 1.6”…
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