Author: VoIP Info
Asterisk cmd Ices
Posted: August 19, 2004 ICES Encode and stream audio using ‘ices’ Description ICES(config.xml) Streams to an icecast server using ices (available separately). A configuration file must be supplied for ices (see contrib/asterisk-ices.xml). ICES() sends the audio output (pcm samples) to stdout. It…
Read More »Asterisk Assistants Verbose Logging
Posted: August 19, 2004 If you find a bug in any of the Asterisk Assistants, you can help fixing it by sending a bug report with a console log of the session in which the error occurred. Here is how to enable…
Read More »Asterisk Assistants Example Log
Posted: August 19, 2004 {CODE()} ===== Thu Aug 19 2004 ===== 15:15:44 Asia/Tokyo ===== 2004-08-19 15:15:49.043 FWD Assistant[8107] “verbose logging is on” 2004-08-19 15:15:49.630 FWD Assistant[8107] “initialising the progress panel window” 2004-08-19 15:15:49.668 FWD Assistant[8107] “preparing to read properties from default configuration”…
Read More »SER module group
Posted: August 19, 2004 group: This module export functions necesarry for group membership checking. There is a database table that contains list of users and groups they belong to. The table is used by functions of this module. A typical use is…
Read More »Asterisk Cisco ATA 186 Emulation
Posted: August 14, 2004 Asterisk Emulation of Cisco ATA 186 Problem: You’re using Asterisk to originate calls to a service (such as 3Tone by L3) which expects to be speaking directly with one or more SIP-based ATA186 adapters. If the service…
Read More »Asterisk Request to schedule in the past
Posted: August 8, 2004 Message:: Request to schedule in the past These messages indicate that something can’t keep up. They occur in tandem with voice prompts becoming very choppy, slow and kind of “under water.” I’ve seen lowering the verbosity, or disconnecting…
Read More »OCADP Analog Telephone Adapter
Posted: August 7, 2004 OCADP Analog Adapter OCADP Analog Telephone Adapter series is one of VoIP terminal products, with one or two RJ45 ports connecting with LAN port, one FXS port connecting with regular PSTN terminal devices like regular phone or PABX…
Read More »Asterisk tips did
Posted: August 7, 2004 Asterisk Howto: Configure your DID lines The telco configures how many digits of the telephone number dialled by the caller is sent down to your PBX. Some connections, PRI or BRI, send all digits and some only the…
Read More »Asterisk tips forking
Posted: August 6, 2004 Forking in the dialplan Q: Daydreams about being able to fork in the dialplan Actually you sort of can – wonder what something like: Dial(Zap/1&LOCAL/s@AlsoDoThis&LOCAL/s@AndThisToo) would do? A: That works. Does not make your CDR any more readable…
Read More »Asterisk Siemens M20
Posted: August 6, 2004 Siemens M20 asterisk howto These are my experiments with the Siemens M20 GSM terminal and Asterisk From Siemens M20 product page The Siemens M20 Terminal is the ideal solution for mobile and stationary industrial applications in telemetry, telematics…
Read More »InfiniRoute Networks
Posted: August 4, 2004 From press release: SAN FRANCISCO, CA – August 3, 2004 – InfiniRoute Networks Inc. (www.infiniroute.com), today announced the general availability of its Voice over Internet Protocol (VoIP) peering service based on the InfiniRoute Managed Voice Platform (MVP). InfiniRoute’s…
Read More »Asterisk cmd WaitMusicOnHold
Posted: August 3, 2004 WaitMusicOnHold Synopsis Wait, playing Music On Hold Description WaitMusicOnHold(delay) Plays hold music specified number of seconds. Deprecated in newer versions. Use MusicOnHold with duration parameter instead. Return code Returns 0 when done, or -1 on hangup. If no…
Read More »Asterisk tips campon
Posted: August 3, 2004 Campon Repeatedly retry a busy extension after 30 sec with music-on-hold in between until that extension answers. exten => 765,1,SetVar(CALLEDEXTEN=${EXTEN}) exten => 765,2,Dial(SIP/20,20,m) exten => 765,3,Goto(campon,camp,1) exten => 765,103,Goto(campon,camp,1) [campon] exten => camp,1,Answer exten => camp,2,Wait(1) exten =>…
Read More »SIPPSTAR
Posted: August 2, 2004 Voice over Internet Protocol (VoIP) means that voice data are not transmitted via telephone cables as they are with conventional telephones but via the computer network. To achieve this, SIPPSTAR uses modern SIP communication protocol and compression methods…
Read More »GSM Codec
Posted: August 1, 2004 GSM (Global System for Mobile communications) is a cellular phone system standard popular outside the USA. GSM includes a codec, often just referred to as the GSM when discussing codecs. The original ‘Full Rate’ GSM speech codec is…
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