simpleopal(1) manual page for SimpleOPAL Version 3.8.2 by Open Phone Abstraction Library on Unix Linux (2.6.32-5-amd64-x86_64)


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)
: 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
: 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:

: Start receiving video immediately.
: Start transmitting video immediately.
: Don't start receiving video immediately.
: 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 headers on outgoing SIP messages
--sip-domain str
: set authentication domain/realm

H.323 options:

-H --no-h323
: Disable H.323 protocol.
: 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.
: 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:, resulting in a call from the pots handset to using the H.323 protocol.
As the pattern field is a regular expression, you could have used in the above .*:26=h323: 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:[email protected]/s
((Please ensure simplopal is the only iax2 app running on your box))