Asterisk cmd BackGroundDetect. Plays a background sound, jumping to an extension on receipt of DTMF tones or to the "talk" extension if it detects talking.BackgroundDetect(filename[|silence[|min[|max]]]) BackgroundDetect is simmilar to Background. It plays back a given filename, waiting for interruption from a given digit (the digit muststart the beginning of a valid extension, or it will be ignored). silence - this is for the silence after you say something.
It is a time in msmin - this is the starting point from which to start to talk. During the playback of the file, audio is monitored in the receive direction, and if a period of non-silence which is greater than 'min' ms yet less than 'max' ms is followed by silence for at least 'silence' ms then the audio playback is aborted and processing jumps to the 'talk' extension if available.
Example [incoming]exten => s,1,Answerexten => s,2,ResponseTimeout(5)exten => s,3,BackgroundDetect(mymenu)exten => s,3,BackgroundDetect(chooseSomething) exten => t,1,Goto(s,2) Interactive Intelligence Blog - Answering Machine Detection Accuracy - Facts and Myths. Asterisk phone snom. Snom phones in general are easy to configure for Asterisk, if both are on the same network.If the phone is behind a NAT, then some care is needed to get best results. Make sure your snom phones have the latest firmware, 3.60x (snom190, end-of-life), 6.5.8 or newer (snom 300/320/360) since STUN support was broken in earlier versions.
Broken Registrar If your Snom phone's firmware has a Support broken Registrar option on the 'SIP line settings' page/tab, this should be set to 'on' for proper phone operation with Asterisk (necessary for Asterisk versions before 1.2).The setting variable for 'Support broken Registrar' is called 'user_sipusername_as_line': Normally the SNOM requires ";line=xyz" to accept an incoming call and be able to map it to the correct line, with the 'broken register' option turned on this may be omitted and the SNOM will still take the call. Note: In firmware 7.x there is now an option available to select "Asterisk" as "Server type". Custom ring tone Alert-Info Ringer: Asterisk, Snom-300 VOIP phone and Power Over Ethernet. - Bend in the Weather.
Well it's coming up to the end of financial year, and like many businesses we needed to make a few purchases. (That or pay more corporate tax -- hmmm let me think.. more toys or more tax? ) Generally we don't make many purchases throughout the year and then buy up large near the end of the financial year.
(well unless something fails!) As such we decided to get a decent VOIP phone. We currently use a LinkSys SPA3102 with our cordless phone, but wanted to get a phone with a decent headset so we could talk hands free easily. We spoke to a few people, did the research and settled on a Snom VOIP phone. You can configure the phone via the LCD menu system, or use the built in web based configuration. Chroot, mount you say?! Whilst this wasn't the sole reason for buying the device, it's handy to know that with a bit of time I could in theory hack the phone.
The Snom phones do, however they use an RJ-11 headset connection instead of a 2.5mm or 3.5mm jack that people are most familar with. Interoperability/PBX/Asterisk - Snom User Wiki. From Snom User Wiki Overview Asterisk is a complete PBX in software. It runs on Linux and provides all of the features you would expect from a PBX and more. Asterisk does voice over IP in three protocols, and can interoperate with almost all standards-based telephony equipment using relatively inexpensive hardware. Asterisk provides Voicemail services with Directory, Call Conferencing, Interactive Voice Response and Call Queuing. It has support for three-way calling, caller ID services, ADSI, SIP and H.323 (as both client and gateway). Asterisk was originally written by Mark Spencer of Digium dba Linux Support Services Inc. All snom phone models can be used with Asterisk. snom Asterisk support snom phones are fully interoperabel with Asterisk. Basic Asterisk configuration The relevant files for SIP phones in Asterisk are sip.conf, extensions.conf and voicemail.conf.The configuration depend on the desired dial plan and usernames e.g. preference to use phone extensions as a usernames. sip.conf.