ASTERWEB Blog

30Mag/16Off

Foto di gruppo a fine “Corso Asterisk 13 Avanzato”

Ringraziamo tutti i partecipanti al corso.

corso-24-26-05-2016

14Mag/16Off

Rilasciato Asterisk 13.9.1

Il giorno 13 maggio 2016, il Team di Sviluppo di Asterisk ha annunciato il rilascio di Asterisk Asterisk 13.9.1.

Dal post originale:

The release of Asterisk 13.9.1 resolves an issue reported by the
community and would have not been possible without your participation.
Thank you!

The following is the issue resolved in this release:

Bugs fixed in this release:
-----------------------------------
* ASTERISK-26007 - res_pjsip: Endpoints deleting early after
upgrade from 13.8.2 to 13.9 (Reported by Greg Siemon)

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.9.1

12Mag/16Off

Funzionalità e servizi (non il prezzo) attirano le PMI verso il VoIP

Secondo la recente ricerca di Frost & Sullivan, nei prossimi cinque anni vedremo un importante numero di piccole e medie aziende adottare soluzioni di telefonia cloud-based o centralini hosted. “Le PMI e le start-up di norma sottoscrivono abbonamenti a piattaforme per le Unified Communications interamente cloud-based o hosted, per i benefici e la flessibilità che esse trasferiscono all’utente”, ha affermato l’analista Wonjae Shim. Secondo le valutazioni dei ricercatori, il mercato dei centralini IP hosted per le PMI nella sola america settentrionale raggiungerà i 350 milioni di dollari nei prossimi 5 anni per giungere al picco di un miliardo di dollari entro il 2021, anno in cui il volume totale del mercato in Europa si attesterà a 17.93 miliardi di dollari nel 2021.

Lo studio di Research and Markets indica che gli “early adopters” di servizi di telefonia IP hosted e UCC in Europa sono primariamente le PMI perché hanno tipicamente budget e staff IT più limitati o competenze non sempre sufficienti per installare e gestire in loco soluzioni avanzate per le telecomunicazioni.

Che il prezzo d’acquisto e il costo operativo totale contino è indubbio, ma secondo Frost & Sullivan non sono l’unico driver della scelta di migrare a centralini VoIP hosted. Secondo gli analisti infatti “le PMI desiderano avvalersi di soluzioni di telefonia ‘as a Service’ perché offrono funzionalità avanzate appositamente sviluppate per migliorare le comunicazioni aziendali e l’operatività interna”.

10Mag/16Off

Rilasciato Asterisk 13.9.0

Il giorno 09 maggio 2016, il Team di Sviluppo di Asterisk ha annunciato il rilascio di Asterisk Asterisk 13.9.0.

Dal post originale:

The release of Asterisk 13.9.0 resolves several issues reported by the community and would have not been possible without your participation.
Thank you!

The following are the issues resolved in this release:

Bugs fixed in this release:
-----------------------------------
* ASTERISK-25963 - func_odbc requires reconnect checks for stale connections (Reported by Ross Beer)
* ASTERISK-25970 - Segfault in pjsip_url_compare (Reported by Dmitriy Serov)
* ASTERISK-25938 - res_odbc: MySQL/MariaDB statement LAST_INSERT_ID() always returns zero. (Reported by Edwin Vandamme)
* ASTERISK-25927 - Removed option "registertrying" is still documented in sip.conf.sample (Reported by Etienne Lessard)
* ASTERISK-25947 - Protocol transfers to stasis applications are missing the StasisStart with the replace_channel object. (Reported by Richard Mudgett)
* ASTERISK-24649 - Pushing of channel into bridge fails; Stasis fails to get app name (Reported by John Bigelow)
* ASTERISK-24782 - StasisEnd event not present for channel that was swapped out for another after completing attended transfer (Reported by John Bigelow)
* ASTERISK-25942 - res_pjsip_caller_id: Transfer results in mixed ConnectedLine information (Reported by George Joseph)
* ASTERISK-25928 - res_pjsip: URI validation done outside of PJSIP thread (Reported by Joshua Colp)
* ASTERISK-25929 - res_pjsip_registrar: AOR_CONTACT_ADDED events not raised (Reported by Joshua Colp)
* ASTERISK-25934 - chan_sip should not require sipregs or updateable sippeers table unless rt (Reported by Jaco Kroon)
* ASTERISK-25888 - Frequent segfaults in function can_ring_entry() of app_queue.c (Reported by Sébastien Couture)
* ASTERISK-25796 - res_pjsip: DOS/Crash when TCP/TLS sockets exceed pjproject PJ_IOQUEUE_MAX_HANDLES (Reported by George Joseph)
* ASTERISK-25707 - Long contact URIs or hostnames can crash pjproject/Asterisk under certain conditions (Reported by George Joseph)
* ASTERISK-25123 - Bracketed IPv6 Contact header parameter unparsable with Asterisk/PJSIP (Reported by Anthony Messina)
* ASTERISK-25874 - app_voicemail: Stack buffer overflow in test_voicemail_notify_endl (Reported by Badalian Vyacheslav)
* ASTERISK-25912 - chan_local passes AST_CONTROL_PVT_CAUSE_CODE without adding them to the local hangupcauses via ast_channel_hangupcause_hash_set (Reported by Jaco Kroon)
* ASTERISK-25885 - res_pjsip: Race condition between adding contact and automatic expiration (Reported by Joshua Colp)
* ASTERISK-25910 - pjproject: Via headers are not parsed when "received" contains an IPv6 address (Reported by George Joseph)
* ASTERISK-25890 - Asterisk 13.8.0 alembic database update fails (Reported by Harley Peters)
* ASTERISK-25894 - [patch] webrtc video broken due to missing marker bits in RTP streams (Reported by Jacek Konieczny)
* ASTERISK-25854 - No audio after HOLD/RESUME - incorrect a=recvonly in SDP from Asterisk (Reported by Robert McGilvray)
* ASTERISK-25873 - res_pjsip: Bundled pjproject: compile error, cannot find -lasteriskpj (Reported by Hans van Eijsden)
* ASTERISK-25882 - ARI: Crash can occur due to race condition when attempting to operate on a hung up channel (Part 2) (Reported by Richard Mudgett)
* ASTERISK-25867 - [patch] Video delay on app_echo (Reported by Jacek Konieczny)
* ASTERISK-24605 - res_parking option parkeddynamic does not work with the core Features 'parkcall' (DTMF initiated parking) (Reported by Philip Correia)
* ASTERISK-25826 - PJSIP / Sorcery slow load from realtime (Reported by Ross Beer)
* ASTERISK-24596 - Unclear how to use Park application with res_parking 'parkeddynamic' enabled. Documentation? (Reported by Philip Correia)
* ASTERISK-24543 - Asterisk 13 responds to SIP Invite with all possible codecs configured for peer as opposed to intersection of configured codecs and offered codecs (Reported by Taylor Hawkes)
* ASTERISK-25825 - Crashes during shutdown when running CLI commands (Reported by Mark Michelson)
* ASTERISK-25407 - Asterisk fails to log to multiple syslog destinations (Reported by Elazar Broad)
* ASTERISK-25510 - [patch]Log to syslog failing (Reported by Michael Newton)
* ASTERISK-25857 - func_aes: incorrect use of strlen() leads to data corruption (Reported by Gianluca Merlo)

Improvements made in this release:
-----------------------------------
* ASTERISK-25865 - Message-Account Missing From PJSIP MWI (Reported by Ross Beer)
* ASTERISK-25444 - [patch]Music On Hold Warning misleading (Reported by Conrad de Wet)

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.9.0