AstManProxy is a multi-threaded proxy server for the Asterisk
Manager Interface. As well as straight proxying, it can also
translate between AMI and HTTP (with output in plaintext, XML,
or CSV formats). SSL is also available (for both AMI and HTTP).
"echo -e" -> "printf"
"exit -1" -> "exit 255"
- some of the patches had hand-rolled chunks to replace /bin/bash
with /bin/sh near a CVS keyword; remove these and replace with a
pre-configure target making it easier to update-patches
- change sample config to disable hardcoded escape sequences for
colours by default
- bump PKGNAME-main
Fixes sscanf without size bounds. The biggest problem affects SIP in
Asterisk 1.6.1+ (i.e. not OpenBSD ports/packages) but the update makes
sense anyway...
just disable by setting the default FLAVOR; the asterisk,h323 entry
in ../Makefile picked it up. the unused pkg/*-speex files don't hurt,
so keep them around. bump PKGNAME (most likely gratuitous, but it's
cheap).
It is completely unmaintained, barely working and prevent from updating the
whole VoIP gang (ptlib, h323plus, opal, gnugk, ekiga) which I'm working on.
some more work still to do but most things should run ok, and it's easier
to handle that in-tree.
KAMAILIO (OpenSER) is a mature and flexible open source SIP server (RFC3261).
It can be used on systems with limitted resources as well as on carrier grade
servers, scaling to up to thousands call setups per second. It is written in
pure C for Unix/Linux-like systems with architecture specific optimizations to
offer high performances. It is customizable, being able to feature as fast load
balancer; SIP server flavours: registrar, location server, proxy server,
redirect server; gateway to SMS/XMPP; or advanced VoIP application server.
where the "pedantic" option is enabled (disabled by default).
Backported rather than updated until I sort out the H323 autoconf
breakage in newer versions.
users with access to the IAX port can use it to verify validity of usernames.
No other code changes in this version.
While there, remove spurious @user from PLIST.
- add -fPIC unconditionally (requested by naddy@) by adding it to
unix.mak (in pwlib) which get sourced by the other ports
feedback from and ok naddy@, thanks!
introduced in 1.4.21 by correcting the order of lock and unlock
in a deadlock avoidance macro... No other changes. Not security,
but if you're running 1.4.21, you definitely want this.
- regen PLIST to remove @bin from a symlink.
split in two: add pkgspec, bump -speex package version
- adjust FULLPKGNAME handling so overrides can be shown clearly
at the top of the Makefile
- add space before assignment operator "FULLPKGNAME$i=" to avoid
potential ambiguity with bad values of $i
speex problem reported by jolan@, thanks!
SpanDSP is a library of DSP functions for telephony, in the 8000 sample
per second world of E1s, T1s, and higher order PCM channels. It contains
low level functions, such as basic filters. It also contains higher
level functions, such as cadenced supervisory tone detection, and a
complete software FAX machine. The software has been designed to avoid
intellectual property issues, using mature techniques where all relevant
patents have expired. See the file DueDiligence for important
information about these intellectual property issues.
from Brad; tested with his work-in-progress CallWeaver.
IAXmodem is a software modem written in C that uses an IAX channel
(commonly provided by an Asterisk PBX system) instead of a traditional
phone line and uses a DSP library instead of DSP hardware chipsets.
IAXmodem was originally conceived to function as a fax modem usable
with HylaFAX, and it does that well. However IAXmodem also has been
known to function with mgetty+sendfax and efax.
.. much cleanup from my earlier verisions, thanks brad@
in RTP codec payload type handling) and AST-2008-003 (SIP channel
can make a call into the context specified in the general section
of sip.conf). Affects all Asterisk users with SIP enabled.
This is a security update only, no changes other than these fixes.
pjsua is an open source command line SIP user agent that is used as
the reference implementation for PJSIP and PJMEDIA. It has many
features, such as:
* Mutiple identities/account registrations
* Concurrent calls and conference (unlimited number, but only up
to 254 sources can be mixed to a single destination)
* Call features: call hold, call transfer (attended or unattended,
with or without refersub).
* SIP Presence/SIMPLE with PIDF and XPIDF support. PUBLISH support.
* Instant messaging and message composing indication
* DTMF digits transmission/receipt (RFC 2833)
* WAV file playing, streaming, and recording.
* Accoustic echo cancellation (AEC).
* Auto-answer, auto-play file, auto-loop RTP
* Support SIP UDP, TCP, and TLS transports. Support for DNS SRV
resolution.
* NAT traversal with rport and STUN.
* Tone generation.
* Codecs: PCMA, PCMU, GSM, Speex (including wideband/16KHz and
ultra-wideband/32KHz), L16 (8-48KHz, mono or stereo), and iLBC.
* Adaptive jitter buffer, adaptive silence detection, and packet
lost concealment audio features.
With lots of testing and help from todd@, sthen@, jakemsr@, jolan@ and
Benny Prijono.
ok todd@ sthen@
fixes an overflow in IMAP voicemail storage reachable by anyone who can
send email to a VM box accessed from the phone. AST-2007-022, found by
sprintf audit.
ok ian@
Net::SIP consists of packages for handling SIP packets, for transport
of packets, for processing packets and on top of all that a simplified
layer for common tasks.
ok sturm@
