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simpleopal(1) [debian man page]

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)
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