Author: VoIP Info
Broadvox
Posted: January 12, 2004 Founded in 2001, Broadvox is a worldwide leader in providing integrated managed VoIP services to SMB, Enterprise and Carrier customers. It has deployed one of the largest, full-featured global VoIP networks and is trusted by more than 160…
Read More »Asterisk cmd AppendCDRUserField
Posted: January 11, 2004 Note: This function is depricated. It has been replaced by Set(CDR(userfield)=Value). Synopsis AppendCDRUserField(Value) Documentation AppendCDRUserField appends data to the CDR, Call Data Record, user field. CDR records are used for billing. Though it is not mentioned in the…
Read More »DDI
Posted: January 10, 2004 If you have point-to-point (PTP) ISDN then you may only connect one ISDN device – the PBX – to the S0 bus. In this case the phone numbers are called DDI instead of MSN (point-to-multipoint, PTMP). Other known…
Read More »Asterisk t extension
Posted: January 9, 2004 t (lower case t): Timeout extension in context. T (capital T): Timeout on AbsoluteTimeout Asterisk standard extensions Asterisk cmd ResponseTimeout Asterisk cmd AbsoluteTimeout: Setting the maximum call time Asterisk config extensions.conf: The dial plan Asterisk | Tips &…
Read More »Asterisk cmd AbsoluteTimeout
Posted: January 9, 2004 Synopsis: Set absolute maximum time of call Status: Deprecated in 1.2 in favor of TIMEOUT(absolute) function Removed in 1.4 Description: AbsoluteTimeout(seconds) Set the absolute maximum amount of time permitted for a call. A setting of 0 disables the…
Read More »Asterisk Call Manager for Windows
Posted: January 6, 2004 by Alex Argov Tikalnetworks Crystal Manager Ver 3.3.5 CrystalManager CrystalManager is a call manager and a personal monitoring tool for Asterisk PBX. Extensions status (BLF) Extensions Grouping Call Pickup Call Transfer Conference Call Double-Click to Call Instant Messaging…
Read More »SIP response class1
Posted: January 4, 2004 SIP response codes, class 1: Provisional messages These are sent within a SIP dialogue 100 Trying 180 Ringing 181 Call Is Being Forwarded 182 Queued 183 Session Progress SIP | SIP response codes | SIP response class1 |…
Read More »SIP Response class5
Posted: January 4, 2004 SIP responses, class 5: Server failures 500 Server Internal Error 501 Not Implemented: The SIP request method is not implemented here 502 Bad Gateway 503 Service Unavailable 504 Server Time-out 505 Version Not Supported: The server does not…
Read More »SIP Response class4
Posted: January 4, 2004 SIP responses, class 4: Request failures 400 Bad Request 401 Unauthorized: Used only by registrars. Proxys should use proxy authorization 407 402 Payment Required (Reserved for future use) 403 Forbidden 404 Not Found: User not found 405 Method…
Read More »SIP Response class3
Posted: January 4, 2004 SIP response codes, class 3xx The 3xx class of responses indicates a redirection of the call 300 Multiple Choices 301 Moved Permanently 302 Moved Temporarily 305 Use Proxy 380 Alternative Service SIP | SIP response codes | SIP…
Read More »SIP Response Codes
Posted: January 4, 2004 The response codes are consistent with, and extend, HTTP/1.1 response codes. Not all HTTP/1.1 response codes are appropriate, and only those that are appropriate are given here. Other HTTP/1.1 response codes should not be used. Also, SIP defines…
Read More »SaRP
Posted: December 29, 2003 SaRP is a SIP and RTP proxy designed specifically to handle the problems inherit in NAT and SIP. The current implementation is written in Perl but a cross-platform C++ version is being worked on. Allow hosts behind a…
Read More »Asterisk disable frame buffer
Posted: December 28, 2003 How to disable the Linux VESA frame buffer Using the X Window System or a VESA Frame Buffer may cause jittery sound. How do you know if you’re using a frame buffer console? Is there a penguin in…
Read More »VoIP
Posted: December 28, 2003 Voice over IP (VoIP) defines a way to carry voice calls over an IP network including the digitization and packetization of the voice streams. IP Telephony utilizes the VoIP standards to create a telephony system where higher level…
Read More »Asterisk tips callback
Posted: December 23, 2003 How to create callback voicemail by Jpiterak Here are a couple of scripts I use to do callback voicemail to my users. This was a request from a customer who was looking at a similar feature on an…
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