VoIP Blog - Page 789
Protocol Verification and Testing
Posted: September 17, 2005 Sources for software and services for testing and verifying that a protocol implementation conforms to a standard. IVR Load Testing Tools BreakingPoint Systems Testing Tools Empirix http://www.empirix.com miTester for SIP: An Open Source SIP Testing tool https://mitesterforsip.sourceforge.net/ GL…
Read More »UIP200
Posted: September 17, 2005 Uniden UIP200 Sip Phone We have a bunch of UIP200s in our office and they’re mostly fine. We use Asterisk for transfers, so we have no issues with call dropping. Two complaints: The time zone info is wrong…
Read More »Asterisk cmd Dictate
Posted: September 17, 2005 Dictate Virtual Dictation Machine: Asterisk now comes with an application to facilitate dictations, see bug 3893. Description Dictate([<base_dir>[|<filename>]]) Start dictation machine using optional base dir for files. Options dial 1 to toggle record and playback modes dial 0…
Read More »Dial Pulse to Touchtone DTMF Converters
Posted: September 17, 2005 Many VOIP devices with FXS ports do not support traditional rotary dial phones. One possible solution (untested) is a Dial pulse to DTMF (Touchtone) converter between the rotary dial phone and the VOIP FXS port Example: Dial Pulse…
Read More »Asterisk cmd BristuffPickUp
Posted: September 16, 2005 PickUp (bristuff) Synopsis Channel independent call pickup. Description PickUp([group]) Tries to pickup the first ringing channel with callgroup == group. If called without the group argument, the pickupgroup of the channel will be used. Related bristuff applications PickUpChan…
Read More »Asterisk func sort
Posted: September 16, 2005 SORT() Synopsis: SORT(key1:val1[…][,keyN:valN]) Description: SORT(key1:val1[…][,keyN:valN]) This function takes a list of keys and values and returns a comma-delimited list of the keys, sorted by each associated value. Values are sorted as floats. Notes The resulting value is…
Read More »Asterisk cmd Cut
Posted: September 16, 2005 Synopsis Cut(newvar=varname,delimiter,fieldspec) Deprecated As of Asterisk 1.2 this command has been depracated. It has been removed as of Asterisk 1.4. Use the function Cut instead. Description Cut(newvar=varname,delimiter,fieldspec) newvar: this variable is set to the result string varname: variable…
Read More »Skype Gateways
Posted: September 16, 2005 A list of products that can be used to connect non-PC devices to the Skype network. Skype itself is available for Windows, Mac OSX, Linux and Pocket-PCs, but for many of the products only windows drivers available. Page…
Read More »Asterisk gui vmail.cgi
Posted: September 16, 2005 Web interface to Voicemail There is a perl script included in the source distribution, vmail.cgi, located in /usr/src/asterisk/contrib/scripts/vmail.cgi Note that ‘make install’ doesn’t install this interface, instead you must run ‘make webvmail‘ separatly. The vmail.cgi script requires that…
Read More »OrderlyQ
Posted: September 16, 2005 OrderlyQ by Orderly Software FREE TRIAL now available to Asterisk users. “50% of all callers hang up after just 45 seconds of waiting on-hold” – Source: Avaya OrderlyQ means your callers never have to wait on hold in…
Read More »VOIP Software
Posted: September 16, 2005 Page Contents Billing Call center monitoring Computer Telephony Integration (CTI) Development Tools Fax Broadcasting IVR Management OSS – Provisioning Queues MNP Solutions Recording Routing Systems Secure Communication Soft Phones SwitchBoards Operator Panels Softswitches Hosted Software Video Conferencing Voice…
Read More »Asterisk cmd PauseQueueMember
Posted: September 16, 2005 Update: in Asterisk 1.4 it has the following details (i.e. you can use an interface rather than an agent): Synopsis Pauses a queue member Description PauseQueueMember([queuename]|interface[|options]): In Asterisk 1.8: PauseQueueMember([queuename],interface,[options],[reason]): Pauses (blocks calls for) a queue member. The…
Read More »QoS
Posted: September 16, 2005 QoS (Quality of Service) is a major issue in VOIP implementations. The issue is how to guarantee that packet traffic for a voice or other media connection will not be delayed or dropped due to interference from other…
Read More »VoIP GSM Gateways
Posted: September 16, 2005 What’s a VoIP GSM Gateway? A VoIP GSM Gateway enables direct routing between IP, digital, analogue and GSM networks. With these devices (fixed cellular terminals) companies can significantly reduce the money they spend on telephony, especially the money…
Read More »RDNIS
Posted: September 15, 2005 RDNIS is Redirected Dialed Number Information Service. RDNIS contains Type of Number (TON), Presentation, and Redirecting Number (RGN), and Reason (busy, no answer, etc.). When redirecting calls, the Originally Called Number (OCN) contains, as its name suggests, the…
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