News from Industry

It’s not yet clear when I can start working on a new-generation Torrus, but here are some nice software projects which would probably inspire the new design, or probably be part of the new design. I haven’t looked into them in depth though.

• Bosun is a distributed monitoring system produced by StackExchange. It uses distributed collector agents which write data into OpenTSDB. Bosun and its collector are written in Go, and OpenTSDB is written in Java.
• InfluxDB is a time-series database written in Go.

and yes, the new project will most probably have its core in Go. But the SNMP discovery engine will most probably remain in Perl because of a big list of supported vendors.

Filed under: Programming Tagged: monitoring, network management, network monitoring, torrus

CEO Mike Leuthner discusses Phonami’s new Admin panel, allowing customers to activate Monster PBX accounts, setup billing, add and remove users directly from Google Apps. 

SendHub co-founder Ryan Pfeffer discusses a product overview, Kazoo at scale, and supporting WebRTC and Mobile.

2600hz and Voxbone discussing WebRTC at WeWork San Francisco, ….sorry for the poor audio quality

2600hz WebRTC Meetup at WeWork, San Francisco, CA from James Solada

Want to become your own carrier!  We’ve launched an incredible new website dedicated to partners, which can be found at partner.2600hz.com. The new site provides information, resources and contact information for VoIP partners interested in Hosted PBX, and soon, Hosted PBX + Mobile.

What does this mean for you? You’ll find new information about our product offerings and how to market immediately and scale quickly.

On our new partner website, you’ll find resources on:

• Hosted PBX – Providing small and medium businesses a fixed-line telecom infrastructure without heavy financial investment. There is no bulky hardware to install or maintain, instead the infrastructure is in the cloud.
• Coming Soon - Hosted PBX + Mobile – We’ve integrated our mobile services with Sprint, providing a native integration with mobile devices.  Our platform will not function as an over-the-top application, instead transforming a cellular phone into a SIP endpoint.

In conjunction with this website launch, we’re introducing a whole bunch of new and exciting features and benefits that include:

• Advanced Functionality - Conferencing, call recording, automatic failover, find-me-follow-me, hunt groups, voicemail-to-email and more. For a complete list of features, visit http://partner.2600hz.com/html/features.html
• White labeling – We provide clients the ability to whitelabel our product as their own and go to market. We also provide white-label documentation including logos, email templates, pricing and restrictions.
• Bring Your Own Device - We enable you to port customer’s existing mobile devices to 2600hz. BYOD gives you the flexibility to fit a customer’s budget by utilizing existing devices.
• Flexible Contracts - We give you the ability to structure fixed PBX, and soon mobile pricing as you see fit. You can offer customers monthly or yearly contracts and set margins with bundled packages.
• Kazoo Integration - Developed by 2600hz, Kazoo employs a cloud-based model designed for cutting edge telecom infrastructures. Kazoo can be deployed in a private cloud environment or can be utilized in a hosted cloud environment.
• Integration - Kazoo is built on over 140 APIs that allow you to integrate with carriers, third-party software vendors, and build your own complex integrations.

Become a 2600hz Partner Today!

Want to take the first step? Contact sales@2600hz.com or sign up today at http://partner.2600hz.com/html/contact.html. We will give you in-depth training and support you as you build your business. Become a valued partner and own the competition!

Also read the press release: http://www.prweb.com/releases/2014/11/prweb12292816.htm

Voxbeam is providing worldwide PSTN connectivity at competitive rates, and it allows you to use any Caller ID, which is very convenient for call forwarding. The Voxbeam gateway authenticates the clients by their IP addresses only, so you need a static IP address, and no username or password are required. The FreeSWITCH configuration shown below allows you to control which destinations should be routed to Voxbeam. With a bit of further extension, you can also control which destinations would use different tariff plans at Voxbeam. This configuration covers only their Standard pricing plan. Here INTERNALDOMAIN is a name of the SIP realm that is used for registered users. We assume that the variable “outbound_caller_id_number” is set elsewhere above in the dialplan.

--- File: ip_profiles/external/voxbeam.xml --- <include>   <gateway name="voxbeam_outbound">     <param name="realm" value="sbc.voxbeam.com" />     <param name="register" value="false" /> <!-- important, so that your caller ID is transmitted properly -->     <param name="caller-id-in-from" value="true"/>   </gateway> </include> --- File: dialplan/INTERNALDOMAIN/05_pstn_outbound.xml --- <include> <!-- Express destination and caller numbers in E.164 notation without leading plus sign. Note that we treat numbers with one leading zero as local Swiss numbers -->   <extension name="pstn_normalize" continue="true">     <condition field="destination_number" expression="^00([1-9]\d+)$" break="never">       <action inline="true" application="set" data="e164_dest=$1"/>     </condition>     <condition field="destination_number" expression="^0([1-9]\d+)$" break="never">       <action inline="true" application="set" data="e164_dest=41$1"/>     </condition>     <condition field="${outbound_caller_id_number}" expression="^00([1-9]\d+)$" break="never">       <action inline="true" application="set" data="e164_cid=$1"/>     </condition>     <condition field="${outbound_caller_id_number}" expression="^0([1-9]\d+)$" break="never">       <action inline="true" application="set" data="e164_cid=41$1"/>     </condition>   </extension> <!-- Here we define that calls to Russia and Ukraine should go through Voxbeam -->   <extension name="pstn_select_itsp" continue="true">     <condition field="${e164_dest}" expression="^(7|38)" break="on-true">       <action inline="true" application="set" data="outbound_itsp=voxbeam"/>     </condition>   </extension>   <!-- send matched calls to Voxbeam -->   <extension name="pstn_voxbeam">     <condition field="${outbound_itsp}" expression="^voxbeam$" break="on-false">       <action application="set" data="effective_caller_id_number=${e164_cid}"/>       <action application="bridge" data="sofia/gateway/voxbeam_outbound/0011103${e164_dest}"/>     </condition>   </extension> <!-- send everything else to Sipcall.ch -->   <extension name="pstn_sipcall">     <condition field="destination_number" expression="^(0\d+)$">       <action application="set" data="effective_caller_id_number=${outbound_caller_id_number}"/>       <action application="bridge" data="sofia/gateway/sipcall/$1"/>     </condition>   </extension> </include>

This is a very simple example, and a bit more logic can be introduced, such as looking up in some kind of a database for least cost routing, and so on.

 

Filed under: Networking Tagged: freeswitch, pbx, sip, voip

The original script is found here: http://www.snip2code.com/Snippet/79027/How-to-install-Go-1-3-in-debian-wheezy

The original script is a bit dated, and now 1.3-1 is the latest version:

## File: go1.3-install-deb.sh
apt-get install devscripts build-essential
apt-get build-dep golang-go

wget http://ftp.de.debian.org/debian/pool/main/g/golang/golang_1.3-3.dsc
wget http://ftp.de.debian.org/debian/pool/main/g/golang/golang_1.3.orig.tar.gz
wget http://ftp.de.debian.org/debian/pool/main/g/golang/golang_1.3-3.debian.tar.xz

dpkg-source -x golang_1.3-3.dsc
cd golang-1.3/
debuild -us -uc
cd ..
dpkg -i \
golang-go_1.3-3_amd64.deb \
golang-src_1.3-3_amd64.deb \
golang-go-linux-amd64_1.3-3_amd64.deb \
vim-syntax-go_1.3-3_all.deb

echo Finished

Filed under: Programming Tagged: go, golang

Drives with problems:

1. KingSpec 16GB (Model Number: KingSpec KSM-mSATA.7i-016MJ, Firmware Revision:  SVN454): quite often, the kernel times out to boot at the start.
2. SuperSSpeed S238: with the old firmware, TRIM operation erased the boot sector. The newer firmware disables TRIM.

Drives without problems (everything works fine with TRIM)

1. MyDigitalSSD Super Boot Drive (Model Number:  SB mSATA SSD, Firmware Revision: S9FM01.8)
2. SanDisk X110 (Model Number: SanDisk SD6SF1M032G1022I, Firmware Revision: X231200)
3. Kingston Now (Model Number: KINGSTON SMS200S330G, Firmware Revision: 524ABBF0)

The testing procedure is quite simple: a background process is massively creating and deleting a small file, and another process calls fstrim every few seconds. Then the health of the filesystem is checked after an hour or so.

while true; do echo xxxxxxxxxxxxxxxxxxxxxxxx >xxx; done & while true; do fstrim -v /; sleep 10; done

 

Filed under: Hardware Tagged: linux, pcengines

PC Engines’ APU board has its mPCIe slot 2 wired to the SIM card socket, which allows using any standard mPCIe 3G modem. Most of modern modems are quite expensive, but there are plenty of Sierra Wireless MC8775 cards at aliexpress.com for around $20 apiece. This is a decent hardware, manufactured around 2007-2011. It doesn’t deliver the highest UMTS speeds possible, but still can be used in situations where speed is unimportant.

The cards that I bought came with firmware version 1_1_8_15, dated 2007/07/17. I didn’t test it fully, but there are some failure reports in the internet.

The firmware upgrade requires an adapter with a SIM card slot. I got mine from this eBay seller.

This page describes the firmware upgrade process. The links to istudioz.net are still valid, but you need to remove # (%23) from the URLs. The 3G watcher for the AirCard 875 is unavailable at its original place, but easy to find with Google. I got mine at this site. The upgrade requires a 32bit Windows machine, and takes about 20 minutes. I upgraded the firmware successfully with my old Vista laptop.

Also I bought the 3G antenna and the pigtail cable at aliexpress.

After inserting the 3G modem into mPCIe slot 2 and booting Debian Wheezy, the device was immediately visible as three serial USB interfaces (/dev/ttyUSB0  /dev/ttyUSB1  /dev/ttyUSB2). ttyUSB0 is used for data, and ttyUSB2 can be used for controlling the device with AT commands. The command “AT^CARDMODE” will tell if the SIM card is inserted, and “AT!GSTATUS?” displays the network status information. “AT+GMR” displays the current firmware version. Ctrl-a Ctrl-x sequence will finish the picocom session.

apt-get install -y wvdial picocom picocom -b 115200 /dev/ttyUSB2 AT^CARDMODE AT!GSTATUS? AT+GMR Ctrl-a Ctrl-x

The following /etc/wvdial.conf works with Sunrise.ch 3G network:

[Dialer Defaults] Modem = /dev/ttyUSB0 Baud = 460800 Init1 = ATZ Init2 = ATQ0 V1 E1 S0=0 &C1 &D2 +FCLASS=0 Phone = *99# Username = '' Password = '' Ask Password = 0 Stupid Mode = 1 Compuserve = 0 Idle Seconds = 0 ISDN = 0 Auto DNS = 1

Execute “wvdial” comand from the command line, and it should immediately connect to the internet. The rest is easy: you can place wvdial into a startup script and execute it automatically at boot time.

Filed under: Networking Tagged: 3G, GSM, linux, networking, pcengines, UMTS

This is a simple test that gives you an estimation of audio conferencing scalability of FreeSWITCH on your hardware.

1. You need one or two FreeSWITCH servers, and one of them should answer to sip:moh@IPADDR:5080. The fastest way is to install this FreeSWITCH configuration: https://github.com/xlab1/voip_qos_probe
2. Edit vars.xml and remove G722 codec (or leave or replace it, if you want to test transcoding performance at the same time).
3. Start FreeSWITCH: service freeswitch start
4. Create conference participants by calling the MOH extension on the remote or the same server. This command will add a few dozens of participants in one go: timeout 2 sh -c "while true; do fs_cli -x 'conference xx dial sofia/internal/moh@IPADDR:5080'; done"
5. check the number of participants: fs_cli -x 'show channels'
6. run “top” or “mpstat -P ALL 1″ to see the CPU load, and add more batches of participants.

This test differs from real world because in a real conference, one speaks and others are listening. Here everyone speaks at the same time. FreeSWITCH evaluates the energy level to find the active speaker before replicating their voice, so I guess the real conference would take less CPU power (need to look into the source code).

Some test results: PC Engines APU platform with 50 conference participants had the CPU usage about 60%. A single core VPS at digitalocean.com was busy at around 50% during a test with 200 participants.

UPD1: (thanks bob bowles) Call out to yourself and monitor the sound quality with your own ear:

fs_cli -x 'conference human dial sofia/external/user@sip.domain.com'
Filed under: Networking Tagged: freeswitch, voip

Cosa sta facendo lo staff ? E gli utenti ?

Ciao a tutti !!

Pubblichiamo il bollettino dello stato del vostro servizio VoIP preferito

Aprile 2014

• Provider in uso 4080 di cui 2450 registrati per ricevere le chiamate
• Interni attivi 1330
• Chiamate 3340 terminate al giorno
• 60 nuovi utenti

Novità:

• Riassunto delle impostazioni per Liberailvoip http://blog.liberailvoip.it/2014/04/16/parametri-di-connessione-liberailvoip/
• Impostazioni avanzate http://blog.liberailvoip.it/2014/04/28/hack-liberailvoip-settings/
• Fix per le chiamate in ingresso per il provider Olimontel
• Fix Bug SSL

Lo staff si sta impegnado su:

• Miglioramento del Monitor
• Correzzione automatica degli indirizzi register dei provider voip inseriti
• Rafforzare la sicurezza dei servizi VoIP
• Applicazione Android per monitorare il proprio account
• Report via mail di eventuali problemi riscontrati sul proprio account
• Supporto ai BLF sui telefoni voip
• Supporto VPN sui telefoni VoIP

Miglioramento del sito web:

• Avviso dello stato dell’account in home page
• Passaggio di tutto il sito in SSL
• DashBoard dello stato globale dei servizi

Approfondimenti

• Aggiornamento Marzo 2014
  

  Cosa sta facendo lo staff ? E gli utenti ?

  Ciao a tutti !!

  Pubblichiamo il bollettino dello stato del vostro servizio VoIP preferito

  Marzo 2014
  • Provider in uso 4150 di cui 2470 registrati per ricevere le [...]
• Aggiornamento Febbraio 2014
  

  Cosa sta facendo lo staff ? E gli utenti ?

  Ciao a tutti !!

  Pubblichiamo il bollettino dello stato del vostro servizio VoIP preferito

  Febbraio 2014
  • Provider in uso 4150 di cui 2440 registrati per ricevere le [...]
• Aggiornamento Gennaio 2014
  

  Cosa sta facendo lo staff ? E gli utenti ?

  Ciao a tutti !!

  Pubblichiamo il bollettino dello stato del vostro servizio VoIP preferito

  Gennaio 2014
  • Provider in uso 4350 di cui 2550 registrati per ricevere le [...]

Questa pagina raccoglie le impostazioni speciali per il servizio LiberIlVoIP

Ciao a tutti !!

Abbiamo deciso di riassumere in questa guida le impostazioni SPECIALI disponibili nella GUI di LiV.

I modificatori vanno aggiunti nella descrizione dell’interno o provider

Interni

1. DRTP: audio p2p (se possibile)
2. QN: disattivazione del ping dello stato dell’interno
3. NNAT: considera l’interno pubblico e non sotto NAT
4. VY: Attiva il supporto alla videochiamata

1 – DRTP

Direct RTP: cerca di eseguire una connessione del flusso audio direttamente tra i due interlocuotri senza passare per il server di LiV. Questa opzione, se supportata dalla rete di connessione, vuole ridurre al minimo la strada percorsa dal flusso audio in modo da avere la maggiore qualità possibile in termini di latenza

2 – QN

Qualify NO: disattiva il controllo continuo della connessione dell’interno.

Settandolo si evita che il server LiV esegua il controllo di raggiungibilità dell’interno, questo comporta un maggior tempo per rilevare la disconnessione dell’interno.

Esempio: se l’interno è impostato con un keepalive di 5min, il server LiV considera l’interno offline solo dopo 5min all’ultimo keepalive lanciato dall’ATA. Quindi se l’ATA viene spento o ci sono problemi di connessione, il server LiV potrebbe considerare l’interno connesso (raggiungibile) anche quando effettivamente non lo è, il chiamante quindi sentirà un prolungato silenzio (decine di secondi) seguito poi dal tono di occupato.

3 – NNAT

No NAT: considera l’interno come se fosse connesso direttamente ad internet (senza NAT)

4 – VY

Video support Yes: Attiva il supporto alla videochiamata sull’interno

 

 

Provider

1. DRTP: audio p2p (se possibile)

1 – DRTP

Direct RTP: cerca di eseguire una connessione del flusso audio direttamente tra i due interlocuotri senza passare per il server di LiV. Se usato con un interno DRTP, il server LiV cercherà di collegare direttamente i flussi RTP tra interno e Provider.

 

 

Se introdurremo altri trik, li pubblicheremo qui.

Approfondimenti

• Parametri di connessione LiberaIlVoIP
  

  Questa pagina raccoglie le impostazioni di connessione per il servizio LiberIlVoIP

  Ciao a tutti !!

  Questa pagina riporta le impostazioni aggiornate e valide per la connessione al servizio VoIP di LiberaIlVoIP.

  Qui saranno elencati [...]

• Riduzione intervallo salvataggio e aggiornamento configurazioni (nei giorni 8-9-10 ottobre)
  

  Insieme al rilascio della nuova piattaforma voip della comunità abbiamo introdotto una nuova modalità di gestione dell’audio voip in “peer to peer”: migliore qualità assicurata a patto di qualche piccolo accorgimento. Per [...]

• Videochiamata con LiberaIlVoIP
  

  Videochiamiamoci con LiV !

  Da oggi LiberaIlVoIP supporta la VIDEOCHIAMATA tra interni

  Ciao a tutti !!
  Siamo liteti di annunciara che da oggi LiberaIlVoIP supporta la videochiamata tra interni !!

  Sono stati provati i seguenti [...]

This is a result of a project where we needed to measure voice QoS parameters (jitter and packet loss) in the customer network. I’ve set up small probe computers (old 10″ Intel Atom netbooks like Acer Aspire One) with FreeSWITCH and a few scripts for test automation. Each test consists of a 30-second call (producing approximately 1500 RTP packets in each direction), and tshark is measuring the received jitter and loss on each side.

Test details and the installation procedure are outlined on Github:

https://github.com/xlab1/voip_qos_probe

 

Filed under: Networking Tagged: freeswitch, network monitoring, sip, voip, xlab1

This test is analogous to the one I described for Intel Atom CPU.This time it’s the new APU board from PC Engines, the maker of famous ALIX and WRAP boards. APU is a fanless appliance board, with a dual-core 1GHz AMD G series CPU. The overall performance is comparable to that of Intel Atom.

In these tests, FreeSWITCH was forwarding the call to itself on request by pressing *1. Each such forwarding resulted in creating four new channels in G722 and G711, thus resulting in transcoding to G711 and back. For example, if “show channels” shows 5 channels, it’s equivalent to 2 simultaneous calls with transcoding.

Test result: 57 channels were running completely fine, 65 channels had slight distortions, and with 85 channels the speech was still recognizable, but with significant distortions. With Speex instead of G722, distortions were quite annoying at 25 channels. Thus, the APU platform can easily be used as a small-to-medium business PBX for  20-30 simultaneous calls if there’s not too much transcoding.

Test details follow.

Debian Wheezy was installed as described in my previous post. Then, GFreeSWITCH version 1.2.23 was installed from packages, as follows:

apt-get install -y curl git sysstat cat >/etc/apt/sources.list.d/freeswitch.list <<EOT deb http://files.freeswitch.org/repo/deb/debian/ wheezy main EOT curl http://files.freeswitch.org/repo/deb/debian/freeswitch_archive_g0.pub | apt-key add - apt-get update apt-get install -y freeswitch-meta-all cd /etc git clone https://github.com/xlab1/freeswitch_conf_minimal.git freeswitch

Then, /etc/freeswitch/dialplan/public/05_test.xml was added as follows:

<include> <!-- Extension 100 accepts the initial call, plays echo, and on pressing *1 it transfers to 101 -->     <extension name="100">       <condition field="destination_number" expression="^100$">         <action application="answer"/>         <action application="bind_meta_app" data="1 a si transfer::101 XML ${context}"/>         <action application="delay_echo" data="1000"/>       </condition>     </extension>     <!-- Extension 101 plays a beep, then makes an outgoing SIP call from our internal profile to our own external profile and extension 200 -->     <extension name="101">       <condition field="destination_number" expression="^101$">         <action application="playback" data="tone_stream://%(100,100,1400,2060,2450,2600)"/>         <action application="unbind_meta_app" data=""/>         <action application="bridge"                 data="{absolute_codec_string=PCMA}sofia/internal/200@${sip_local_network_addr}:5080"/>       </condition>     </extension> <!-- Extension 200 returns the call to 100 as a new outgoing SIP call from our internal profile to our own external profile -->     <extension name="200">       <condition field="destination_number" expression="^200$">         <action application="answer"/>         <action application="bridge"                 data="{max_forwards=65}{absolute_codec_string=G722}sofia/internal/100@${sip_local_network_addr}:5080"/>       </condition>     </extension>     </include>

After sending the initial call from a SIP phone to extension 100 at our APU’s IP address and port 5080, after pressing *1 we get 2 new channels with transcoding. Below are results of “mpstat -P ALL 1″ command during the test:

# quite clear sound root@apu:/etc/freeswitch# fs_cli -x 'show channels' | grep total 57 total. 11:35:07 PM  CPU    %usr   %nice    %sys %iowait    %irq   %soft  %steal  %guest   %idle 11:35:08 PM  all   41.71    0.00    8.00    0.00    0.00    0.57    0.00    0.00   49.71 11:35:08 PM    0   43.68    0.00    5.75    0.00    0.00    1.15    0.00    0.00   49.43 11:35:08 PM    1   40.45    0.00   10.11    0.00    0.00    0.00    0.00    0.00   49.44 # slight distortions root@apu:/etc/freeswitch# fs_cli -x 'show channels' | grep total 65 total. 11:36:27 PM  CPU    %usr   %nice    %sys %iowait    %irq   %soft  %steal  %guest   %idle 11:36:28 PM  all   55.98    0.00    8.70    0.00    0.00    0.54    0.00    0.00   34.78 11:36:28 PM    0   55.91    0.00    7.53    0.00    0.00    2.15    0.00    0.00   34.41 11:36:28 PM    1   55.43    0.00    9.78    0.00    0.00    0.00    0.00    0.00   34.78 # significant distortions root@apu:/etc/freeswitch# fs_cli -x 'show channels' | grep total 85 total. 11:37:34 PM  CPU    %usr   %nice    %sys %iowait    %irq   %soft  %steal  %guest   %idle 11:37:35 PM  all   71.13    0.00    9.28    0.00    0.00    2.06    0.00    0.00   17.53 11:37:35 PM    0   71.72    0.00    9.09    0.00    0.00    2.02    0.00    0.00   17.17 11:37:35 PM    1   71.58    0.00    9.47    0.00    0.00    2.11    0.00    0.00   16.84

If G722 is replaced with Speex codec, the CPU load is significantly higher, and already with 25 channels the distortions are quite significant:

# speex 8kHz, distortions root@apu:/etc/freeswitch# fs_cli -x 'show channels' | grep total 25 total. 12:59:46 AM  CPU    %usr   %nice    %sys %iowait    %irq   %soft  %steal  %guest   %idle 12:59:47 AM  all   54.10    0.00    1.64    0.00    0.00    0.00    0.00    0.00   44.26 12:59:47 AM    0   53.85    0.00    2.20    0.00    0.00    0.00    0.00    0.00   43.96 12:59:47 AM    1   54.95    0.00    1.10    0.00    0.00    0.00    0.00    0.00   43.96
Filed under: Networking Tagged: freeswitch, pbx, pcengines, sip, voip

Questa pagina raccoglie le impostazioni di connessione per il servizio LiberIlVoIP

Ciao a tutti !!

Questa pagina riporta le impostazioni aggiornate e valide per la connessione al servizio VoIP di LiberaIlVoIP.

Qui saranno elencati i parametri di connessione SIP di LiV sempre aggiornati:

Server di registrazione (registrar/server/SIP server): sip.liberailvoip.it -> 94.23.65.208
NOTA: Usare l’ip al posto del dns SOLO SE STRETTAMENTE necesario, se usate l’ip e poi un giorno non si registra PRIMA di postare NON FUNZIONA, controlla l’ip indicato in questa discussione.

Porte di registrazione: 53 80 5060-5065
NOTA: Usare porte diverse dalla 5060 solo se STRETTAMENTE necessario, cioè solo se con 5060 non si registra a causa di blocchi dell’ISP o router/NAT

Protocollo di registrazione: UDP, TCP
NOTA: Usare TCP se con UDP non ricevi le chiamate. TCP è attivato in via sperimentale.

Codec attualmente attivi: ulaw,alaw,gsm,ilbc,g722,g726,g726aal2,g723,g729
NOTA: se imposti g729:
Inband DTMF is not supported on codec g729. Use RFC2833

Proxy sip (outbound proxy): sip.liberailvoip.it
NOTA: Non serve quasi mai con LiV, quindi solo se ci sono motivazioni particolari va impostato

Stun Server: stun.liberailvoip.it
NOTA: Non serve quasi mai con LiV, quindi solo se ci sono motivazioni particolari va impostato

Cosiglio di usare i DNS di opendns
208.67.222.222
208.67.220.220

 

Se hai problemi a ricevere le chiamate in ingresso e quindi non funziona nemmeno il Test di chiamata, prova ad usare il protocollo TCP invece dell’UDP.

Approfondimenti

• Hack LiberaIlVoIP Settings
  

  Questa pagina raccoglie le impostazioni speciali per il servizio LiberIlVoIP

  Ciao a tutti !!

  Abbiamo deciso di riassumere in questa guida le impostazioni SPECIALI disponibili nella GUI di LiV.

  I modificatori vanno aggiunti nella [...]

• Facciamoci un multicentrum: connessione mobile flat giornaliera a 25 cent (e le novità in arrivo da H3G)
  

  Tim sforna una nuova tariffa “alle vitamine” pensata per i giovani.Uno scatto giornaliero di 25 centesimi e 1000 connessioni WAP gratuite (senza limite di traffico).Sfruttiamola per la navigazione web in mobilità.Una flat [...]

• Grandstream GXP 2020: parametri per l’ Italia
  

  Come “da tradizione” completiamo la recensione dell’hardware appena testato (il Grandstream GXP 2020) fornendo i parametri necessari per il perfetto funzionamento dell’apparecchio su linee italiane. Nel caso specifico, trattandosi di un telefono [...]