[Pkg-voip-commits] r8586 - in /opal/trunk/debian: changelog control libopal3.8.1.install libopal3.8.2.install rules simpleopal.1
msp at alioth.debian.org
msp at alioth.debian.org
Sun Jul 25 09:02:50 UTC 2010
Author: msp
Date: Sun Jul 25 09:02:44 2010
New Revision: 8586
URL: http://svn.debian.org/wsvn/pkg-voip/?sc=1&rev=8586
Log:
* New Upstream Release
* Refresh simpleopal.1
Added:
opal/trunk/debian/libopal3.8.2.install
- copied unchanged from r8435, opal/trunk/debian/libopal3.8.1.install
Removed:
opal/trunk/debian/libopal3.8.1.install
Modified:
opal/trunk/debian/changelog
opal/trunk/debian/control
opal/trunk/debian/rules
opal/trunk/debian/simpleopal.1
Modified: opal/trunk/debian/changelog
URL: http://svn.debian.org/wsvn/pkg-voip/opal/trunk/debian/changelog?rev=8586&op=diff
==============================================================================
--- opal/trunk/debian/changelog (original)
+++ opal/trunk/debian/changelog Sun Jul 25 09:02:44 2010
@@ -1,3 +1,10 @@
+opal (3.8.2~dfsg-1) experimental; urgency=low
+
+ * New Upstream Release
+ * Refresh simpleopal.1
+
+ -- Mark Purcell <msp at debian.org> Sun, 25 Jul 2010 18:53:53 +1000
+
opal (3.8.1~dfsg-1) experimental; urgency=low
* New upstream release
Modified: opal/trunk/debian/control
URL: http://svn.debian.org/wsvn/pkg-voip/opal/trunk/debian/control?rev=8586&op=diff
==============================================================================
--- opal/trunk/debian/control (original)
+++ opal/trunk/debian/control Sun Jul 25 09:02:44 2010
@@ -10,14 +10,14 @@
libtheora-dev, libgsm1-dev, libspeex-dev, libspeexdsp-dev,
libcapi20-dev
Build-Conflicts: libopal-dev, libx264-dev,
- libopal3.8.1, libopal3.6.8, libopal3.6.6
+ libopal3.8.2, libopal3.6.8
Standards-Version: 3.8.3
Homepage: http://www.opalvoip.org/
Vcs-Svn: svn://svn.debian.org/pkg-voip/opal/trunk/
Vcs-Browser: http://svn.debian.org/wsvn/pkg-voip/opal/?op=log
DM-Upload-Allowed: yes
-Package: libopal3.8.1
+Package: libopal3.8.2
Architecture: any
Depends: ${shlibs:Depends}, ${misc:Depends}, libspeex1, libspeexdsp1
Description: Open Phone Abstraction Library - successor of OpenH323
@@ -35,7 +35,7 @@
Section: libdevel
Architecture: any
Conflicts: openmpi-dev
-Depends: ${misc:Depends}, libopal3.8.1 (= ${binary:Version}), libpt-dev (>= 2.8.1), libspeexdsp-dev,
+Depends: ${misc:Depends}, libopal3.8.2 (= ${binary:Version}), libpt-dev (>= 2.8.1), libspeexdsp-dev,
libsrtp0-dev [alpha amd64 armel hppa i386 mips mipsel powerpc s390]
Suggests: pkg-config
Recommends: libopal-doc
@@ -50,7 +50,7 @@
Package: simpleopal
Section: comm
Architecture: any
-Depends: ${shlibs:Depends}, ${misc:Depends}, libopal3.8.1 (= ${binary:Version})
+Depends: ${shlibs:Depends}, ${misc:Depends}, libopal3.8.2 (= ${binary:Version})
Description: Simple example from the OPAL project
This package contains a small H323 and SIP client given as an example. You
can find its code on the doc package.
@@ -63,7 +63,7 @@
Package: libopal-dbg
Section: debug
Priority: extra
-Depends: ${misc:Depends}, libopal3.8.1 (= ${binary:Version})
+Depends: ${misc:Depends}, libopal3.8.2 (= ${binary:Version})
Conflicts: libopal3.6.8-dbg, libopal3.6.6-dbg
Replaces: libopal3.6.8-dbg, libopal3.6.6-dbg
Architecture: any
Modified: opal/trunk/debian/rules
URL: http://svn.debian.org/wsvn/pkg-voip/opal/trunk/debian/rules?rev=8586&op=diff
==============================================================================
--- opal/trunk/debian/rules (original)
+++ opal/trunk/debian/rules Sun Jul 25 09:02:44 2010
@@ -22,9 +22,8 @@
dh_auto_build -- opt docs
$(MAKE) PTLIBDIR=/usr OPALDIR=$(CURDIR) -C samples/simple
-# Upstream dropped ChangeLog :-(
-#override_dh_installchangelogs:
-# dh_installchangelogs ChangeLog*.txt
+override_dh_installchangelogs:
+ dh_installchangelogs ChangeLog*.txt
override_dh_auto_test:
if (samples/simple/obj*/simpleopal --help >/dev/null);then \
Modified: opal/trunk/debian/simpleopal.1
URL: http://svn.debian.org/wsvn/pkg-voip/opal/trunk/debian/simpleopal.1?rev=8586&op=diff
==============================================================================
--- opal/trunk/debian/simpleopal.1 (original)
+++ opal/trunk/debian/simpleopal.1 Sun Jul 25 09:02:44 2010
@@ -1,17 +1,300 @@
-.\" Written by Eugen Dedu, Eugen.Dedu at pu-pm.univ-fcomte.fr
-.\"
-.TH simpleopal 1 2008-09-21 GNU simpleopal
+.\" 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 \- simple SIP and H323 client
-.SH SYNOPSIS
-.B simpleopal \fR[\fIoptions\fR] \fB-l
-.br
-.B simpleopal \fR[\fIoptions\fR] [\fIalias@\fR]\fBhostname\fR (no gatekeeper)
-.br
-.B simpleopal \fR[\fIoptions\fR] \fBalias\fR[\fI at hostname\fR] (with gatekeeper)
+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
-Small SIP and H323 client.
-.SH OPTIONS
-Please execute \fBsimpleopal \-\-help\fR for help.
-.SH NOTES
-This program is one of the examples given in the OPAL library.
+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 at 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:<dn2ip>
+pots:.* = sip:<da>
+pc:.* = sip:<da>
+.IP
+If SIP & IAX2 are not enabled and H.323 is enabled:
+.IP
+pots:.*\e*.*\e*.* = h323:<dn2ip>
+pots:.* = h323:<da>
+pc:.* = h323:<da>
+.IP
+If POTS is enabled:
+.IP
+h323:.* = pots:<dn>
+sip:.* = pots:<dn>
+iax2:.* = pots:<dn>
+.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 at misery.digium.com/s
+.IP
+((Please ensure simplopal is the only iax2 app running on your box))
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