The remserial program acts as a communications bridge between a
TCP/IP network port and a Linux device such as a serial port. Any
character-oriented Linux /dev device will work.
The program can also use pseudo-ttys as the device. A pseudo-tty
is like a serial port in that it has a /dev entry that can be opened
by a program that expects a serial port device, except that instead
of belonging to a physical serial device, the data can be intercepted
by another program. The remserial program uses this to connect a
network port to the "master" (programming) side of the pseudo-tty
allowing the device driver (slave) side to be used by some program
expecting a serial port. See example 3 below for details.
The program can operate as a server accepting network connections
from other machines, or as a client, connecting to remote machine
that is running the remserial program or some other program that
accepts a raw network connection. The network connection passes
data as-is, there is no control protocol over the network socket.
Multiple copies of the program can run on the same computer at the
same time assuming each is using a different network port and
device.
pkglint -r --network --only "migrate"
As a side-effect of migrating the homepages, pkglint also fixed a few
indentations in unrelated lines. These and the new homepages have been
checked manually.
pkglint --only "https instead of http" -r -F
With manual adjustments afterwards since pkglint 19.4.4 fixed a few
indentations in unrelated lines.
This mainly affects projects hosted at SourceForce, as well as
freedesktop.org, CTAN and GNU.
pkglint --only "https instead of http" -r -F
With manual adjustments afterwards since pkglint 19.4.4 fixed a few
indentations in unrelated lines.
This mainly affects projects hosted at SourceForce, as well as
freedesktop.org, CTAN and GNU.
Changes:
1.41.0
------
[-] * Documentation improvements.
[-] * Updated MySQL script to be compatible with current server versions.
[-] * Fixed SMSD operation on phones with more SMS folders.
[-] * Fixed off by one in Python example script.
[-] * Fixed PostgreSQL compilation on openSUSE.
[-] * Several compatibility fixes with recent compilers.
[-] * Improved USSD support.
[-] * Localization updates.
Changelog:
16.7.0
Security bugs fixed in this release:
-----------------------------------
[ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter Addr of a peer (Reported by Andrey V. T.)
[ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows system commands execution (Reported by Eliel Sardañons)
Improvements made in this release:
-----------------------------------
[ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached (Reported by Daniel)
[ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam Banks)
[ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when format differs from attachfmt column (Reported by cmaj)
[ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any .lock files in the voice mail directory on startup. (Reported by Michael)
[ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold re-invites (Reported by Torrey Searle)
[ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by Florian Floimair)
Bugs fixed in this release:
-----------------------------------
[ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
[ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0 (Reported by George Joseph)
[ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream if codecs create additional streams and offer does not have them (Reported by nappsoft)
[ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR points to a not existent AOR (Reported by Ross Beer)
[ASTERISK-28644] - Stale comment in app_queue about ring_entry exception (Reported by Walter Doekes)
[ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on hangup when TEST_FRAMEWORK enabled (Reported by Bernhard Schmidt)
[ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check failure when negociated ptime is not default ptime. (Reported by Frederic LE FOLL)
[ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the same (Reported by Ross Beer)
[ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received (Reported by Salah Ahmed)
[ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported by Kevin Harwell)
[ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported by Michael Cargile)
[ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a short alarm (Reported by Frederic LE FOLL)
[ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet length doesn't match (Reported by Joshua Elson)
[ASTERISK-26481] - FILE function grabs garbage along with read data when target line has no newline (Reported by Jonathan Harris)
[ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix bridge (Reported by Kevin Harwell)
[ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by Joshua C. Colp)
[ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
[ASTERISK-28572] - Memory leaks in res_calendar_exchange and res_calendar_icalendar (Reported by Yoooooo Ha)
[ASTERISK-28585] - ari/resource_events: Crash in event session cleanup (Reported by Kevin Harwell)
[ASTERISK-28590] - utils.c throws repeated warnings; "pthread_attr_setstacksize: Invalid argument" (Reported by Speed Dial Dave)
[ASTERISK-28578] - race condition on pjsip channelstats command (Reported by Salah Ahmed)
[ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column (Reported by Christoph Moench-Tegeder)
[ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
[ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported by Niklas Larsson)
[ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
[ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi container (Reported by Kevin Harwell)
[ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported by Marian Piater)
[ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
[ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI (Reported by Jeremiah Gadd)
[ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by Jørgen H)
[ASTERISK-28534] - Segmentation fault when there is no priority for an extension (Reported by Timothy Vanderaerden)
[ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured (Reported by Juan Martin)
[ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
[ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
[ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C. Colp)
[ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD (Reported by Guido Falsi)
[ASTERISK-23756] - setvar directive when used in template and a child of said template, results in duplicate variable names (Reported by Michael Goryainov)
New Features made in this release:
-----------------------------------
[ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead of only "sending" (Reported by lvl)
[ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported by Martin Tomec)
[ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
[ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI (Reported by Jeremiah Gadd)
[ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by Jørgen H)
[ASTERISK-28534] - Segmentation fault when there is no priority for an extension (Reported by Timothy Vanderaerden)
[ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured (Reported by Juan Martin)
[ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
[ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
[ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C. Colp)
[ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD (Reported by Guido Falsi)
[ASTERISK-23756] - setvar directive when used in template and a child of said template, results in duplicate variable names (Reported by Michael Goryainov)
New Features made in this release:
-----------------------------------
[ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead of only "sending" (Reported by lvl)
[ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported by Martin Tomec)
[ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization (Reported by Joshua C. Colp)
16.6.0
Security bugs fixed in this release:
-----------------------------------
[ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream causes crash (Reported by Alexei Gradinari)
Bugs fixed in this release:
-----------------------------------
[ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
[ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
[ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C. Colp)
[ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD (Reported by Guido Falsi)
[ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from SLIN16 to SLIN32 (Reported by Ruddy G)
[ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel hangs up (Reported by Frederic LE FOLL)
[ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in cdr.conf (Reported by Frederic LE FOLL)
[ASTERISK-28499] - translate: Crash when frame does not have a "src" field set (Reported by Gregory Massel)
[ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end of a struct (Reported by Alexander Traud)
[ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by Chris Savinovich)
[ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC only supports up to 32 characters (Reported by Dan Cropp)
[ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not checking mailstream (Reported by Alexei Gradinari)
[ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
[ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp->session to srtp_protect/unprotect causing SEGV (Reported by Jonas Swiatek)
[ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua C. Colp)
[ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by Salah Ahmed)
[ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries (Reported by Ian Jones)
[ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
[ASTERISK-28477] - Crash when not specifying "dbfile" in res_config_sqlite3.conf (Reported by Dennis)
[ASTERISK-28478] - Crash performing "core reload" with modified res_config_sqlite3.conf (Reported by Dennis)
[ASTERISK-26968] - chan_pjsip: Transfer() does not result in TRANSFERSTATUS reflecting SIP response to transfer (Reported by Dan Cropp)
[ASTERISK-28282] - AST_SCHED_REPLACE_UNREF causes wait-on-self deadlocks (in chan_sip) (Reported by Walter Doekes)
New Features made in this release:
-----------------------------------
[ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron Clark)
[ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From header URI domain (Reported by Stas Kobzar)
AUTOFIX: Makefile:290: Replacing "${PKGSRC_COMPILER} == \"clang\"" with "${PKGSRC_COMPILER:Mclang}".
The PKGSRC_COMPILER can be a list of chained compilers, e.g. "ccache
distcc clang". Therefore, comparing it using == or != leads to wrong
results in these cases.
Qodem is a from-scratch clone implementation of the Qmodem
communications program made popular in the days when Bulletin Board
Systems ruled the night. Qodem emulates the dialing directory and the
terminal screen features of Qmodem over both modem and Internet
connections.
OK kamil@
Asterisk is a complete PBX in software. It provides all of the
features you would expect from a PBX and more. Asterisk does voice
over IP in three protocols, and can interoperate with almost all
standards-based telephony equipment using relatively inexpensive
hardware.
Asterisk provides Voicemail services with Directory, Call Conferencing,
Interactive Voice Response, Call Queuing. It has support for
three-way calling, caller ID services, ADSI, SIP and H.323 (as both
client and gateway).
From the website:
2009-06-22 15:13:28 Version .30 released. FOP2 is born.
I have just released FOP 0.30, this version works reasonably well with
Asterisk 1.6. There are no new features. It is a maintenance and
compatiblity release.
I would also like to inform you that FOP2 is born. It is the next
generation FOP. A complete rewrite focused on the user and taking into
account all what I learned throughout the years.
Please visit http://www.fop2.com to read more about it.
FOP1 will not be discontinued. I will keep mantaining it but I won't be
adding any new features. I will fix bugs and make it work with future
asterisk releases.
