Asterisk 1.6.2.13 Now Available (1.6.2.12 Re-Release)
Il Team di Sviluppo di Asterisk ha annunciato il rilascio di Asterisk 1.6.2.13
Dal post originale:
This release resolves an issue where the .version and ChangeLog files were not
updated for 1.6.2.12. Asterisk 1.6.2.13 has no additional changes from 1.6.2.12
other than the .version, ChangeLog and summary files.
Questo il hangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.13
Asterisk 1.6.2.12 Now Available
Il Team di Sviluppo di Asterisk ha annunciato il rilascio di Asterisk 1.6.2.12
Dal post originale:
The release of Asterisk 1.6.2.12 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following is a sample of the issues resolved in this release:
* Fix issue where DNID does not get cleared on a new call when using
immediate=yes with ISDN signaling.
(Closes issue #17568. Reported by wuwu. Patched by rmudgett)
* Several updates to res_config_ldap.
(Closes issue #13573. Reported by navkumar. Patched by navkumar, bencer.
Tested by suretec)
* Prevent loss of Caller ID information set on local channel after masquerade.
(Closes issue #17138. Reported by kobaz, patched by jpeeler)
* Fix SIP peers memory leak.
(Closes issue #17774. Reported, patched by kkm)
* Add Danish support to say.conf.sample
(Closes issue #17836. Reported, patched by RoadKill)
* Ensure SSRC is changed when media source is changed to resolve audio delay.
(Closes issue #17404. Reported, tested by sdolloff. Patched by jpeeler)
* Only do magic pickup when notifycid is enabled.
A new way of doing BLF pickup was introduced into 1.6.2. This feature adds a
call-id value into the XML of a SIP_NOTIFY message sent to alert a subscriber
that a device is ringing. This option should only be enabled when the new
'notifycid' option is set, but this was not the case. Instead the call-id
value was included for every RINGING Notify message, which caused a
regression for people who used other methods for call pickup.
(Closes issue #17633. Reported, patched by urosh. Patched by dvossel.
Tested by: dvossel, urosh, okrief, alecdavis)
Questo il ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.12
Asterisk 1.4.36 Now Available
Il Team di Sviluppo di Asterisk ha annunciato il rilascio di Asterisk 1.4.36
Dal post originale:
The release of Asterisk 1.4.36 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following is a sample of the issues resolved in this release candidate:
* Fix issue where DNID does not get cleared on a new call when using
immediate=yes with ISDN signaling.
(Closes issue #17568. Reported by wuwu. Patched by rmudgett)
* Fix issue where SIP promiscuous redirect could fail to dial the
redirect (app_queue).
* Fixes issue with translator frame not getting freed. This issue prevented
G.729 licenses from being freed up.
(Closes issue #17630. Reported by manvirr. Patched by dvossel)
* Ensure SSRC is changed when media source is changed to resolve audio delay.
(Closes issue #17404. Reported, tested by sdolloff. Patched by jpeeler)
* Q931 - Sending PROGRESS after sending ALERTING is a protocol error.
(Closes issue #17874. Reported, patched by nic_bellamy)
Questo il ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.4.36
Altri rilasci del Team di Sviluppo di Asterisk di luglio
Asterisk 1.6.2.10 Now Available
Asterisk 1.4.35-rc1 and 1.6.2.11-rc1 Now Available
Asterisk 1.6.2.11-rc2 Now Available
libpri 1.4.12-beta1 Now Available
Saluti a tutti !
Asterisk 1.8.0 beta
Il 23 e il 27 luglio sono state rilasciate le beta 1 e 2 di Asterisk 1.8.0.
Vi ricordo che la 1.8 è una LTS (Long Term Support) come la 1.4 (tabella releases).
Questi i link dei post originali:
Asterisk 1.8.0-beta1 Now Available
Asterisk 1.8.0-beta2 Now Available
Cominaciamo a sbirciare e a sperare 🙂
Saluti e buone ferie a tutti !
libpri 1.4.11.3 Now Available
Il Team di Sviluppo di Asterisk ha annunciato il rilascio della versione 14.11.3 di LIBPRI (la 1.4.11.2 è stata rilasciata meno di un mese fa).
Questo il contenuto del post originale:
This release fixes a regression in the calling number assignment logic:
- Calling Number assignment logic change in libpri 1.4.11. Restored the old
behaviour if there is more than one calling number in the incoming SETUP
message. A network provided number is reported as ANI.
(Closes issue #17495. Reported and tested by ibercom. Patched by rmudgett)
For a full list of changes in the current release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/libpri/releases/ChangeLog-1.4.11.3
Thank you for your continued support of Asterisk!
Asterisk 1.4.33.1 Released
Il Team di Sviluppo di Asterisk ha annunciato il rilascio di Asterisk 1.4.33.1
Dal post originale:
The release of Asterisk 1.4.33.1 resolves a regression involving the use of FXO signaling in chan_dahdi where a channel could continue ringing after it has been answered.
For a full list of changes in the current release, please see the
ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.4.33.1
Thank you for your continued support of Asterisk!
P.S. il post era del 22 giugno, ma lo avevo lasciato tra le bozze . Sorry !
Configurazione iptables (integra il post fail2ban)
www.asterweb.org
A seguito di alcune richieste pervenuteci, ad integrazione del post fail2ban, abbiamo realizzato un tutorial per avere una configurazione iptables funzionante ed immediatamente utilizzabile. Questo il link del tutorial
Ovviamente per chi volesse meglio comprendere il funzionamento di iptables, si rimanda ad un più approfondito studio.
Saluti
Asterweb
Elastix nel suo blog (finalmente) consiglia fail2ban
Elastix
Piano piano, ci sono arrivati anche loro.
Con un post del 26 giugno, infatti, rimandano ad un altro post (fatto da altri) dove viene spiegato come installare e configurare fail2ban per proteggere il proprio pbx.
Domandona: perché non lo mettono di default su Elastix, come tra l'altro fa da sempre pbxinaflash ?.
Detto ciò, vi rimando al tutorial che avevo fatto a suo tempo e che spiega come installare e configurare file2ban.
A proposito, per chi non lo sapesse, fail2ban si preoccupa di controllare i file di log generati dai vari programmi (ssh, asterisk, etc) e se verifica tentativi di accesso (ad esempio per SSH) o di registrazione di client SIP non andati a buon fine (per Asterisk) banna l'ip modificando iptables.
SIPVicious ora si protegge da se stesso
www.asterweb.org
La versione 0.2.6 del "famigerato" SIPVicious contiene il nuovo tool svcrash.py che è in grado di interrompere eventuali attacchi fatti con svwar o svcrack, mandandoli in crash (almeno quelli delle versioni precedenti).
Non ho ancora avuto modo di testarne pienamente il funzionamento. Una volta testato farò un altro post.
Alla prossima.
Asterisk 1.4.33 Now Available
Il Team di Sviluppo di Asterisk ha annunciato il rilascio di Asterisk 1.4.33
Dal post originale:
The release of Asterisk 1.4.33 resolves several issues reported by the
community, and would have not been possible without your participation.
Thank you!
The following are a few of the issues resolved by community developers:
- Remove arbitrary size limitation for hints
(Closes issue #17257. Reported, patched by tim_ringenbach) - Fix incorrectly typed indications for [nz] stutter and dialrecall
(Closes issue #17359. Reported, patched by alecdavis) - Make AgentComplete message more consistent
(Closes issue #15638. Reported, patched by elbriga) - Missing fallback to audio fax feature when T.38 re-INVITE failed
(Closes issue #16692. Reported, patched by vrban) - Don't hang up on a queue caller if the file we attempt to play does not exist
(Closes issue #17061. Reported by RoadKill)
For a full list of changes in the current release, please see the
ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.4.33
Thank you for your continued support of Asterisk!
Asterisk 1.6.2.9 Now Available
Il Team di Sviluppo di Asterisk ha annunciato il rilascio di Asterisk 1.6.2.9
Dal post originale:
The release of Asterisk 1.6.2.9 resolves several issues reported by the
community, and would have not been possible without your participation.
Thank you!
The following are a few of the issues resolved by community developers:
- Fix the PickupChan() application
(Closes issue #16863. Reported, patched by schern. Patched by cjacobsen.
Tested by Graber, cjacobsen, lathama, rickead2000, dvossel) - Improve logging by displaying line number
(Closes issue #16303. Reported by dant. Patched by pabelanger. Tested by
dant, pabelanger, lmadsen) - Notify CLI when modules are loaded/unloaded
(Closes issue #17308. Reported, patched by pabelanger. Tested by russell) - Make the Makefile logic more explicit and move the Snow Leopard logic down to
where it's not executed on non-Darwin systems
(Closes issue #17028. Reported by pabelanger. Patched by seanbright,
tilghman. Tested by pabelanger) - Manager cookies are not compatible with RFC2109. Make that no longer true.
(Closes issue #17231. Reported, patched by ecarruda) - With IMAP backend, messages in INBOX were counted twice for MWI
(Closes issue #17135. Reported by edhorton. Patched by ebroad, tilghman) - Fix possible segfault when logging
(Closes issue #17331. Reported, patched by under. Patched by dvossel) - Fix memory hogging behavior of app_queue
(Closes issue #17081. Reported by wliegel. Patched by mmichelson) - Allow type=user SIP endpoints to be loaded properly from realtime
(Closes issue #16021. Reported, patched by Guggemand)
Additionally, the following issue may be of interest:
- Fix transcode_via_sln option with SIP calls and improve PLC usage
(Review: https://reviewboard.asterisk.org/r/622/)
For a full list of changes in the current release, please see the
ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.9
Thank you for your continued support of Asterisk!
libpri 1.4.11.2 Now Available
Il Team di Sviluppo di Asterisk ha annunciato il rilascio della versione 14.11.2 di LIBPRI.
Questo il contenuto del post originale:
This release fixes situation where Q.SIG calling name in FACILITY message was
not reported to the upper layer:
* pri_facility.c: Q.SIG calling name in FACILITY message not
reported to the upper layer. Q.SIG can send the CallingName,
CalledName, and ConnectedName in stand alone FACILITY messages.
If the CallingName was not sent in the SETUP message, the caller
id name was not reported to the upper layer.
(Closes issue #17458. Reported, tested by: jsmith. Patched by
rmudgett)
For a full list of changes in the current release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/libpri/releases/ChangeLog-1.4.11.2
Thank you for your continued support of Asterisk!
Asterisk 1.6.2.8 Now Available
Il Team di Sviluppo di Asterisk ha annunciato il rilascio di Asterisk 1.6.2.8
Dal post originale:
The following are a few of the issues resolved by community developers:
* Enable auto complete for CLI command 'logger set level'.
(Closes issue #17152. Reported, patched by pabelanger)
* Make the mixmonitor thread process audio frames faster.
(Closes issue #17078. Reported, tested by geoff2010. Patched by dhubbard)
* Add missing 'useragent' field to sip-friends.sql file.
(Closes issue #17171. Reported, patched by thehar)
* Add example dialplan for dialing ISN numbers (http://www.freenum.org)
(Closes issue #17058. Reported, patched by pprindeville)
* Fix issue with double "sip:" in header field.
(Closes issue #15847. Reported, patched by ebroad)
* Add ability to generate ASCII documentation from the TeX files by running 'make asterisk.txt'.
(Closes issue #17220. Reported by lmadsen. Tested, patched by pabelanger)
* When StopMonitor() is called, ensure that it will not be restarted by a channel event.
(Closes issue #16590. Reported, patched by kkm)
* Small error in the T.140 RTP port verbose log.
(Closes issue #16998. Reported, patched by frawd. Tested by russell)
For a full list of changes in the current release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.8
Thank you for your continued support of Asterisk!
Asterisk 1.4.32 Now Available
Il Team di Sviluppo di Asterisk ha annunciato il rilascio di Asterisk 1.4.32.
Dal post originale:
The following are a few of the issues resolved by community developers:
* Make the mixmonitor thread process audio frames faster.
(Closes issue #17078. Reported, tested by: geoff2010. Patched by dhubbard)
* When StopMonitor is called, ensure that it will not be restarted by a channel event.
(Closes issue #16590. Reported, patched by: kkm)
* Fix up hidecallerid feature in chan_dahdi.
(Closes issue #17143, #7321. Reported, patched by djenson99)
* Resolve deadlocks in chan_local.
(Closes issue #17185. Reported, tested by schmoozecom, GameGamer43)
* Ensure channel state is not incorrectly set in the case of a very early answer.
(Closes issue #17067. Reported, patched by tzafrir)
* Registration fix for SIP realtime.
(Closes issue #17266. Reported, patched by Nick_Lewis. Tested by sberney)
For a full list of changes in the current release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.4.32
Thank you for your continued support of Asterisk!