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Asterisk 1.4.17 (Current branch)
01-25-2008
Asterisk is a hybrid TDM and packet voice PBX (Private Branch eXchange) and IVR platform with ACD functionality. It acts as middleware between the Internet (IAX, SIP, MGCP, Skinny, H.323), telephony channels (like Zaptel, T1, PRI, E1, FXO, FXS, VoIP, VoFR, ISDN, modems, Internet Phone Jack, etc.), and applications (like voice-mail, conferencing, directories, MP3 players, intercoms, etc.). It has many advanced features such as a codec translation API. The base distribution includes several channel backends, as well as applications. However, the beauty of Asterisk is its ability to be extended using its APIs, dynamic module loader, and AGI scripting interface. End users can even write their own applications that run on the system in C or any scripting language of their choice. License: GNU General Public License (GPL) Changes:|
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SIMPLEOPAL(1) User Commands SIMPLEOPAL(1) NAME
SimpleOPAL - manual page for SimpleOPAL Version 3.8.2 by Open Phone Abstraction Library on Unix Linux (2.6.32-5-amd64-x86_64) DESCRIPTION
SimpleOPAL Version 3.8.2 by Open Phone Abstraction Library on Unix Linux (2.6.32-5-amd64-x86_64) Usage : [options] -l : [options] [alias@]hostname (no gatekeeper) : [options] alias[@hostname] (with gatekeeper) General options: -l --listen : Listen for incoming calls. -d --dial-peer spec : Set dial peer for routing calls (see below) --no-std-dial-peer : Do not include the standard dial peers -a --auto-answer : Automatically answer incoming calls. -u --user name : Set local alias name(s) (defaults to login name). -p --password pwd : Set password for user (gk or SIP authorisation). -D --disable media : Disable the specified codec (may be used multiple times) -P --prefer media : Prefer the specified codec (may be used multiple times) -O --option fmt:opt=val : Set codec option (may be used multiple times) : fmt is name of codec, eg "H.261" : opt is name of option, eg "Target Bit Rate" : val is value of option, eg "48000" --srcep ep : Set the source endpoint to use for making calls --disableui : disable the user interface Audio options: -j --jitter [min-]max : Set minimum (optional) and maximum jitter buffer (in milliseconds). -e --silence : Disable transmitter silence detection. Video options: --rx-video : Start receiving video immediately. --tx-video : Start transmitting video immediately. --no-rx-video : Don't start receiving video immediately. --no-tx-video : Don't start transmitting video immediately. --grabber dev : Set the video grabber device. --grabdriver dev : Set the video grabber driver (if device name is ambiguous). --grabchannel num : Set the video grabber device channel. --display dev : Set the video display device. --displaydriver dev : Set the video display driver (if device name is ambiguous). --video-size size : Set the size of the video for all video formats, use : "qcif", "cif", WxH etc --video-rate rate : Set the frame rate of video for all video formats --video-bitrate rate : Set the bit rate for all video formats -C string : Enable and select video rate control algorithm SIP options: -I --no-sip : Disable SIP protocol. -r --register-sip host : Register with SIP server. --sip-proxy url : SIP proxy information, may be just a host name : or full URL eg sip:user:pwd@host --sip-listen iface : Interface/port(s) to listen for SIP requests : '*' is all interfaces, (default udp$:*:5060) --sip-user-agent name: SIP UserAgent name to use. --sip-ui type : Set type of user indications to use for SIP. Can be one of 'rfc2833', 'info-tone', 'info-string'. --use-long-mime : Use long MIME headers on outgoing SIP messages --sip-domain str : set authentication domain/realm H.323 options: -H --no-h323 : Disable H.323 protocol. --no-h323s : Do not create secure H.323 endpoint -g --gatekeeper host : Specify gatekeeper host, '*' indicates broadcast discovery. -G --gk-id name : Specify gatekeeper identifier. --h323s-gk host : Specify gatekeeper host for secure H.323 endpoint -R --require-gatekeeper : Exit if gatekeeper discovery fails. --gk-token str : Set gatekeeper security token OID. --disable-grq : Do not send GRQ when registering with GK -b --bandwidth bps : Limit bandwidth usage to bps bits/second. -f --fast-disable : Disable fast start. -T --h245tunneldisable : Disable H245 tunnelling. --h323-listen iface : Interface/port(s) to listen for H.323 requests --h323s-listen iface : Interface/port(s) to listen for secure H.323 requests : '*' is all interfaces, (default tcp$:*:1720) Line Interface options: -L --no-lid : Do not use line interface device. --lid device : Select line interface device (eg Quicknet:013A17C2, default *:*). --country code : Select country to use for LID (eg "US", "au" or "+61"). Sound card options: -S --no-sound : Do not use sound input/output device. -s --sound device : Select sound input/output device. --sound-in device : Select sound input device. --sound-out device : Select sound output device. IVR options: -V --no-ivr : Disable IVR. -x --vxml file : Set vxml file to use for IVR. --tts engine : Set the text to speech engine IP options: --translate ip : Set external IP address if masqueraded --portbase n : Set TCP/UDP/RTP port base --portmax n : Set TCP/UDP/RTP port max --tcp-base n : Set TCP port base (default 0) --tcp-max n : Set TCP port max (default base+99) --udp-base n : Set UDP port base (default 6000) --udp-max n : Set UDP port max (default base+199) --rtp-base n : Set RTP port base (default 5000) --rtp-max n : Set RTP port max (default base+199) --rtp-tos n : Set RTP packet IP TOS bits to n --stun server : Set STUN server Debug options: -t --trace : Enable trace, use multiple times for more detail. -o --output : File for trace output, default is stderr. -X --no-iax2 : Remove support for iax2 -h --help : This help message. Dial peer specification: General form is pattern=destination where pattern is a regular expression matching the incoming calls destination address and will translate it to the outgoing destination address for making an outgoing call. For example, picking up a PhoneJACK handset and dialling 2, 6 would result in an address of "pots:26" which would then be matched against, say, a spec of pots:26=h323:10.0.1.1, resulting in a call from the pots handset to 10.0.1.1 using the H.323 protocol. As the pattern field is a regular expression, you could have used in the above .*:26=h323:10.0.1.1 to achieve the same result with the addition that an incoming call from a SIP client would also be routed to the H.323 client. Note that the pattern has an implicit ^ and $ at the beginning and end of the regular expression. So it must match the entire address. If the specification is of the form @filename, then the file is read with each line consisting of a pattern=destination dial peer specification. Lines without and equal sign or beginning with '#' are ignored. The standard dial peers that will be included are: If SIP is enabled but H.323 & IAX2 are disabled: pots:.**.**.* = sip:<dn2ip> pots:.* = sip:<da> pc:.* = sip:<da> If SIP & IAX2 are not enabled and H.323 is enabled: pots:.**.**.* = h323:<dn2ip> pots:.* = h323:<da> pc:.* = h323:<da> If POTS is enabled: h323:.* = pots:<dn> sip:.* = pots:<dn> iax2:.* = pots:<dn> If POTS is not enabled and the PC sound system is enabled: iax2:.* = pc: h323:.* = pc: sip:. * = pc: If IVR is enabled then a # from any protocol will route it it, ie: .*:# = ivr: If IAX2 is enabled then you can make a iax2 call with a command like: simpleopal -I -H iax2:guest@misery.digium.com/s ((Please ensure simplopal is the only iax2 app running on your box)) SimpleOPAL Version 3.8.2 by Open Phone Abstraction Library on Unix LiJuly(201032-5-amd64-x86_64) SIMPLEOPAL(1)