Rilasciato Kamailio V5.3.7
Kamailio SIP Server v5.3.7 stable è uscito - una versione minore che include correzioni nel codice e nella documentazione dalla v5.3.6. La compatibilità del file di configurazione e dello schema del database viene mantenuta, il che significa che non è necessario modificare nulla per l'aggiornamento.
The link of the post... :https://www.kamailio.org/w/2020/11/kamailio-v5-3-7-released/
Rilasciato Kamailio V5.4.2
Kamailio SIP Server v5.4.2 stable è uscito - una versione minore che include correzioni nel codice e nella documentazione dalla v5.2.7. La compatibilità del file di configurazione e dello schema del database viene mantenuta, il che significa che non è necessario modificare nulla per l'aggiornamento.
The link of the post... : https://www.kamailio.org/w/2020/10/kamailio-v5-4-2-released/
Rilasciato Kamailio v5.2.7
Kamailio SIP Server v5.2.7 stable è uscito - una versione minore che include correzioni nel codice e nella documentazione dalla v5.2.6. La compatibilità del file di configurazione e dello schema del database viene mantenuta, il che significa che non è necessario modificare nulla per l'aggiornamento.
... the link of the post: https://www.kamailio.org/w/2020/05/kamailio-v5-2-7-released/
Voip al galoppo, nel 2020 lo userà il 95% delle imprese
Voip al galoppo, nel 2020 lo userà il 95% delle imprese
Secondo l’indagine condotta da Uli oggi il Voice Over Ip è adottato dal 52% degli utenti italiani. A spingere la svolta “digital” il caro-prezzi dell’analogico.
Nel 2020 soltanto il 5% dei privati userà il fisso, mentre il 95% delle imprese ricorrerà sistematicamente alla trasmissione della voce via dati...
... the link of the post: https://www.corrierecomunicazioni.it/telco/voip-al-galoppo-nel-2020-lo-usera-il-95-delle-imprese/
Asterisk 13.30.0, 16.7.0, and 17.1.0 Now Available
Asterisk 13.30.0, 16.7.0, and 17.1.0 Now Available
Il TEAM di Asterisk ha annunciato di aver rilasciato il download di Asterisk 13.30.0, 16.7.0 e 17.1.0.
Le versioni sono disponibili per il download immediato all'indirizzo: https://downloads.asterisk.org/pub/telephony/asterisk/
... the link of the post: https://community.asterisk.org/t/asterisk-13-30-0-16-7-0-and-17-1-0-now-available/82066
Kamailio-tests, a testing framework for Kamailio developers
Kamailio-tests, a testing framework for Kamailio developers
Giacomo Vaca has published a detailed article about Kamailio Testing Framework:
https://www.giacomovacca.com/2019/11/kamailio-tests-testing-framework-for.html
... the link of the post: :https://www.kamailio.org/w/2019/11/kamailio-testing-framework/
The Many Business Productivity Advantages of VoIP
The Many Business Productivity Advantages of VoIP
VoIP has become a ubiquitous yet extremely important part of evolving communication systems, and is key to achieving maximum business productivity.
... the link of the post: https://www.tmcnet.com/channels/business-voip/articles/443704-many-business-productivity-advantages-voip.htm
Asterisk 17.0.0-rc1 Now Available
The Asterisk Development Team would like to announce the first release candidate of Asterisk 17.0.0.
This release candidate is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 17.0.0-rc1 resolves several issues reported by the
community.
...the link of the post: https://www.asterisk.org/downloads/asterisk-news/asterisk-1700-rc1-now-available
Asterisk 1.6.2.15 Now Available
Il giorno 8 dicembre, il Team di Sviluppo di Asterisk ha annunciato il rilascio della beta Asterisk 1.6.2.15.
Dal post originale:
he release of Asterisk 1.6.2.15 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:
* When using chan_skinny, don't crash when parking a non-bridged call.
(Closes issue #17680. Reported, tested by jmhunter. Patched, tested by DEA)
* Add ability for Asterisk to try both the encoded and unencoded subscription
URI for a match in hints.
(Closes issue #17785. Reported, tested by ramonpeek. Patched by tilghman)
* Set the caller id on CDRs when it is set on the parent channel.
(Closes issue #17569. Reported, patched by tbelder)
* Ensure user portion of SIP URI matches dialplan when using encoded characters
(Closes issue #17892. Reported by wdoekes. Patched by jpeeler)
* Resolve issue where Party A in an analog 3-way call would continue to hear
ringback after party C answers.
(Patched by rmudgett)
* Fix problem with qualify option packets for realtime peers never stopping.
The option packets not only never stopped, but if a realtime peer was not in
the peer list multiple options dialogs could accumulate over time.
(Closes issue #16382. Reported by lftsy. Tested by zerohalo. Patched by
jpeeler)
* Multiple fixes related to Local channels.
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.15
Asterisk 1.8.1-rc1 Now Available
Il giorno 23 novembre, il Team di Sviluppo di Asterisk ha annunciato il rilascio della beta Asterisk 1.8.1-rc1.
The release of Asterisk 1.8.1-rc1 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 when using directmedia. Asterisk needs to limit the codecs offered
to just the ones that both sides recognize, otherwise they may end up sending
audio that the other side doesn't understand.
(Closes issue #17403. Reported, patched by one47. Tested by one47, falves11)
* Resolve issue where Party A in an analog 3-way call would continue to hear
ringback after party C answers.
(Patched by rmudgett)
* Fix playback failure when using IAX with the timerfd module.
(Closes issue #18110. Reported, tested by tpanton. Patched by jpeeler)
* Fix problem with qualify option packets for realtime peers never stopping.
The option packets not only never stopped, but if a realtime peer was not in
the peer list multiple options dialogs could accumulate over time.
(Closes issue #16382. Reported by lftsy. Tested by zerohalo. Patched by
jpeeler)
* Fix issue where it is possible to crash Asterisk by feeding the curl engine
invalid data.
(Closes issue #18161. Reported by wdoekes. Patched by tilghman)
For a full list of changes in this release candidate, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.1-rc1
Thank you for your continued support of Asterisk!
Asterisk 1.6.2.15-rc1 Now Available
Il giorno 23 novembre, il Team di Sviluppo di Asterisk ha annunciato il rilascio della beta Asterisk 1.6.2.15-rc1.
Dal post originale:
The release of Asterisk 1.6.2.15-rc1 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:
* When using chan_skinny, don't crash when parking a non-bridged call.
(Closes issue #17680. Reported, tested by jmhunter. Patched, tested by DEA)
* Add ability for Asterisk to try both the encoded and unencoded subscription
URI for a match in hints.
(Closes issue #17785. Reported, tested by ramonpeek. Patched by tilghman)
* Set the caller id on CDRs when it is set on the parent channel.
(Closes issue #17569. Reported, patched by tbelder)
* Ensure user portion of SIP URI matches dialplan when using encoded characters
(Closes issue #17892. Reported by wdoekes. Patched by jpeeler)
* Resolve issue where Party A in an analog 3-way call would continue to hear
ringback after party C answers.
(Patched by rmudgett)
* Fix problem with qualify option packets for realtime peers never stopping.
The option packets not only never stopped, but if a realtime peer was not in
the peer list multiple options dialogs could accumulate over time.
(Closes issue #16382. Reported by lftsy. Tested by zerohalo. Patched by
jpeeler)
* Multiple fixes related to Local channels.
For a full list of changes in this release candidate, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.15-rc1
Thank you for your continued support of Asterisk!
Asterisk 1.8.0 Release Candidate 3 Now Available
Il Team di Sviluppo di Asterisk ha annunciato il rilascio della terza beta della 1.8.0.
Dal post originale:
This release candidate contains fixes since the release candidate as reported by
the community. A sampling of the changes in this release candidate include:
* Still build chan_sip even if res_crypto cannot be built (use, but not depend)
(Reported by a user on the mailing list. Patched by tilghman)
* Get notifications for call files only when a file is closed, not when created
(Closes issue #17924. Reported by mkeuter. Patched by abeldeck)
* Fixes to chan_gtalk to allow outbound DTMF support to work correctly. Gtalk
expects the DTMF to arrive on the RTP stream and not via jingle DTMF
signalling.
(Patched by dvossel. Tested by malcolmd)
* Fixes to allow chan_gtalk to communicate with the Gmail web client.
(Patched by phsultan and dvossel)
* Fix to GET DATA to allow audio to be streamed via an AGI.
(Closes issue #18001. Reported by jamicque. Patched by tilghman)
* Resolve dnsmgr memory corruption in chan_iax2.
(Closes issue #17902. Reported by afried. Patched by russell, dvossel)
A short list of available features includes:
* Secure RTP
* IPv6 Support in the SIP channel driver
* Connected Party Identification Support
* Calendaring Integration
* A new call logging system, Channel Event Logging (CEL)
* Distributed Device State using Jabber/XMPP PubSub
* Call Completion Supplementary Services support
* Advice of Charge support
* Much, much more!
A full list of new features can be found in the CHANGES file.
http://svn.digium.com/view/asterisk/branches/1.8/CHANGES?view=checkout
For a full list of changes in the current release candidate, please see the
ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.0-rc3
Elastix 2.0: Primi screenshots e download versione test
Elastix
Il 31 dicembre sul sito di Elastix è stato pubblicato il post che annuncia il rilascio della versione Alfa (in test) di Elastix 2.0 e che mostra alcuni screenshots della nuova release.
Ecco i link per il donwload delle versioni test: 32 bit e 64 bit
Screenshots
Dashboard
Elastix 2.0 - Dashboard 1
Distribuzioni Asterisk: Trixbox, Elastix e PBXInaFlash. Quali differenze ?
www.asterweb.org
Iniziamo subito ad analizzare le 3 distribuzioni di cui sopra:
- tutte e 3 utilizzano LAMP (Linux, Apache, MySql e PHP)
- Linux: il Sistema Operativo e in specifico centOS 5
- Apache: il Web Server e in specifico Apache 2
- MySql: come gestore di DB e in specifico MySql 5
- Php: come linguaggio interpetrato di programmazione Web e in specifico Php5
D: Ma allora: quali sono le differenze ?
R: Per la parte PBX nessuna ! Sono proprio identici !
Le differenze sono negli strumenti che mettono a disposizione per la configurazione del sistema (inteso proprio come "macchina" e non come PBX) e per la gestione dei programmi/pacchetti con l'utilizzo di propri repository.
Ecco qui una sintetica tabella comparativa:
| Servizio/Funzione da Web GUI | Trixbox | Elastix | PBXInaFlash |
| Configurazione rete | Si | Si | No |
| Gestione pacchetti | Si | Si | No |
| Configurazione smtp | Si | Si | No |
| Gestione Server di posta | No | Si | No |
| Gestione Webmail | No | Si Round Cube |
No |
| Hylafax/Iaxmodem | No | Si | No |
| Gestione Sistema (Webmin) | No | No | Si |
Asterisk CLI – comandi su database interno
www.asterweb.org
Vediamo oggi il "Database Interno" di Asterisk, che si base su DB Berkeley.
La struttura prevede il raggruppamento dei dati in "famiglie" e l'associazione delle stesse a chiavi univoche.
Alcuni dei comandi per gestire il DB, sono:
- database show
- database put
- database del
Vediamoli in dettaglio:
PBX-shell*CLI> database show
Visualizza tutti i record del DB.
Asterisk CLI - database show
PBX-shell*CLI> database show [family]
Visualizza tutti i record del DB che appartengono a quella famiglia. Es. database show CW
Asterisk CLI - database show
PBX-shell*CLI> database put [family] [key] [valore]
Il comando consente di inserire un record nel DB. Es. database put CW 307 DISABLED
Asterisk CLI - database put
PBX-shell*CLI> database del [family] [key]
Il comando consente di eliminare uno o più record che soddisfano la condizione. Es. database del CW 307