Provided by: simpleopal_3.10.10~dfsg-2.1ubuntu3_amd64 bug

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 AbstrJulyo2010brary on Unix Linux (2.6.32-5-aSIMPLEOPAL(1)