.\" DO NOT MODIFY THIS FILE! It was generated by help2man 1.38.2. .TH SIMPLEOPAL "1" "July 2010" "SimpleOPAL Version 3.8.2 by Open Phone Abstraction Library on Unix Linux (2.6.32-5-amd64-x86_64)" "User Commands" .SH 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) .SH DESCRIPTION SimpleOPAL Version 3.8.2 by Open Phone Abstraction Library on Unix Linux (2.6.32\-5\-amd64\-x86_64) .PP Usage : [options] \fB\-l\fR .TP : [options] [alias@]hostname (no gatekeeper) .TP : [options] alias[@hostname] (with gatekeeper) .SS "General options:" .TP \fB\-l\fR \fB\-\-listen\fR : Listen for incoming calls. .TP \fB\-d\fR \fB\-\-dial\-peer\fR spec : Set dial peer for routing calls (see below) .TP \fB\-\-no\-std\-dial\-peer\fR : Do not include the standard dial peers .TP \fB\-a\fR \fB\-\-auto\-answer\fR : Automatically answer incoming calls. .TP \fB\-u\fR \fB\-\-user\fR name : Set local alias name(s) (defaults to login name). .TP \fB\-p\fR \fB\-\-password\fR pwd : Set password for user (gk or SIP authorisation). .TP \fB\-D\fR \fB\-\-disable\fR media : Disable the specified codec (may be used multiple times) .TP \fB\-P\fR \fB\-\-prefer\fR media : Prefer the specified codec (may be used multiple times) .TP \fB\-O\fR \fB\-\-option\fR 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" .TP \fB\-\-srcep\fR ep : Set the source endpoint to use for making calls .TP \fB\-\-disableui\fR : disable the user interface .SS "Audio options:" .TP \fB\-j\fR \fB\-\-jitter\fR [min\-]max : Set minimum (optional) and maximum jitter buffer (in milliseconds). .TP \fB\-e\fR \fB\-\-silence\fR : Disable transmitter silence detection. .SS "Video options:" .TP \fB\-\-rx\-video\fR : Start receiving video immediately. .TP \fB\-\-tx\-video\fR : Start transmitting video immediately. .TP \fB\-\-no\-rx\-video\fR : Don't start receiving video immediately. .TP \fB\-\-no\-tx\-video\fR : Don't start transmitting video immediately. .TP \fB\-\-grabber\fR dev : Set the video grabber device. .TP \fB\-\-grabdriver\fR dev : Set the video grabber driver (if device name is ambiguous). .TP \fB\-\-grabchannel\fR num : Set the video grabber device channel. .TP \fB\-\-display\fR dev : Set the video display device. .TP \fB\-\-displaydriver\fR dev : Set the video display driver (if device name is ambiguous). .TP \fB\-\-video\-size\fR size : Set the size of the video for all video formats, use : "qcif", "cif", WxH etc .TP \fB\-\-video\-rate\fR rate : Set the frame rate of video for all video formats .HP \fB\-\-video\-bitrate\fR rate : Set the bit rate for all video formats .TP \fB\-C\fR string : Enable and select video rate control algorithm .SS "SIP options:" .TP \fB\-I\fR \fB\-\-no\-sip\fR : Disable SIP protocol. .TP \fB\-r\fR \fB\-\-register\-sip\fR host : Register with SIP server. .TP \fB\-\-sip\-proxy\fR url : SIP proxy information, may be just a host name : or full URL eg sip:user:pwd@host .TP \fB\-\-sip\-listen\fR iface : Interface/port(s) to listen for SIP requests : '*' is all interfaces, (default udp$:*:5060) .HP \fB\-\-sip\-user\-agent\fR name: SIP UserAgent name to use. .TP \fB\-\-sip\-ui\fR type : Set type of user indications to use for SIP. Can be one of 'rfc2833', 'info\-tone', 'info\-string'. .TP \fB\-\-use\-long\-mime\fR : Use long MIME headers on outgoing SIP messages .TP \fB\-\-sip\-domain\fR str : set authentication domain/realm .SS "H.323 options:" .TP \fB\-H\fR \fB\-\-no\-h323\fR : Disable H.323 protocol. .TP \fB\-\-no\-h323s\fR : Do not create secure H.323 endpoint .TP \fB\-g\fR \fB\-\-gatekeeper\fR host : Specify gatekeeper host, '*' indicates broadcast discovery. .TP \fB\-G\fR \fB\-\-gk\-id\fR name : Specify gatekeeper identifier. .TP \fB\-\-h323s\-gk\fR host : Specify gatekeeper host for secure H.323 endpoint .HP \fB\-R\fR \fB\-\-require\-gatekeeper\fR : Exit if gatekeeper discovery fails. .TP \fB\-\-gk\-token\fR str : Set gatekeeper security token OID. .TP \fB\-\-disable\-grq\fR : Do not send GRQ when registering with GK .TP \fB\-b\fR \fB\-\-bandwidth\fR bps : Limit bandwidth usage to bps bits/second. .TP \fB\-f\fR \fB\-\-fast\-disable\fR : Disable fast start. .TP \fB\-T\fR \fB\-\-h245tunneldisable\fR : Disable H245 tunnelling. .TP \fB\-\-h323\-listen\fR iface : Interface/port(s) to listen for H.323 requests .TP \fB\-\-h323s\-listen\fR iface : Interface/port(s) to listen for secure H.323 requests : '*' is all interfaces, (default tcp$:*:1720) .SS "Line Interface options:" .TP \fB\-L\fR \fB\-\-no\-lid\fR : Do not use line interface device. .TP \fB\-\-lid\fR device : Select line interface device (eg Quicknet:013A17C2, default *:*). .TP \fB\-\-country\fR code : Select country to use for LID (eg "US", "au" or "+61"). .SS "Sound card options:" .TP \fB\-S\fR \fB\-\-no\-sound\fR : Do not use sound input/output device. .TP \fB\-s\fR \fB\-\-sound\fR device : Select sound input/output device. .TP \fB\-\-sound\-in\fR device : Select sound input device. .TP \fB\-\-sound\-out\fR device : Select sound output device. .SS "IVR options:" .TP \fB\-V\fR \fB\-\-no\-ivr\fR : Disable IVR. .TP \fB\-x\fR \fB\-\-vxml\fR file : Set vxml file to use for IVR. .TP \fB\-\-tts\fR engine : Set the text to speech engine .SS "IP options:" .TP \fB\-\-translate\fR ip : Set external IP address if masqueraded .TP \fB\-\-portbase\fR n : Set TCP/UDP/RTP port base .TP \fB\-\-portmax\fR n : Set TCP/UDP/RTP port max .TP \fB\-\-tcp\-base\fR n : Set TCP port base (default 0) .TP \fB\-\-tcp\-max\fR n : Set TCP port max (default base+99) .TP \fB\-\-udp\-base\fR n : Set UDP port base (default 6000) .TP \fB\-\-udp\-max\fR n : Set UDP port max (default base+199) .TP \fB\-\-rtp\-base\fR n : Set RTP port base (default 5000) .TP \fB\-\-rtp\-max\fR n : Set RTP port max (default base+199) .TP \fB\-\-rtp\-tos\fR n : Set RTP packet IP TOS bits to n .TP \fB\-\-stun\fR server : Set STUN server .SS "Debug options:" .TP \fB\-t\fR \fB\-\-trace\fR : Enable trace, use multiple times for more detail. .TP \fB\-o\fR \fB\-\-output\fR : File for trace output, default is stderr. .TP \fB\-X\fR \fB\-\-no\-iax2\fR : Remove support for iax2 .TP \fB\-h\fR \fB\-\-help\fR : This help message. .SS "Dial peer specification:" .IP 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. .IP 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. .IP Note that the pattern has an implicit ^ and $ at the beginning and end of the regular expression. So it must match the entire address. .IP 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. .IP The standard dial peers that will be included are: .IP If SIP is enabled but H.323 & IAX2 are disabled: .IP pots:.*\e*.*\e*.* = sip: pots:.* = sip: pc:.* = sip: .IP If SIP & IAX2 are not enabled and H.323 is enabled: .IP pots:.*\e*.*\e*.* = h323: pots:.* = h323: pc:.* = h323: .IP If POTS is enabled: .IP h323:.* = pots: sip:.* = pots: iax2:.* = pots: .IP If POTS is not enabled and the PC sound system is enabled: .IP iax2:.* = pc: h323:.* = pc: sip:. * = pc: .IP If IVR is enabled then a # from any protocol will route it it, ie: .TP \&.*:# = ivr: .IP If IAX2 is enabled then you can make a iax2 call with a command like: .TP simpleopal \fB\-I\fR \fB\-H\fR iax2:guest@misery.digium.com/s .IP ((Please ensure simplopal is the only iax2 app running on your box))