---
name: "simpleopal"
section: "1"
platform: "linux"
package: "simpleopal"
version: "3.10.10~dfsg2-2+b2"
release: "Debian 9 (Stretch)"
platforms: ["linux"]
url: "https://new.carta.tech/simpleopal"
---

# 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)

## 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))