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))