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voip-info.org
Welcome to the VOIP Wiki - a reference guide to all things VOIP
This Wiki covers everything related to VOIP, software, hardware, service providers, reviews, configurations, standards, tips & tricks and everything else related to voice over IP networks, IP telephony and Internet Telephony. However, the Wiki is primarily for information, not for advertising.
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Your contributions are welcome, please read the How to add information to this wiki page and the Posting Guidelines before you post.
NEWS
2010-03-12 - Asterisk Open Source Conference coming up...
2010-03-12 - Cisco claims CRS-3 IP Routing will revolutionize the internet...
2010-03-12 - VoIPon Brings the Best of CeBIT to You VoIPon Interviews Sangoma, AVM, Pika, Patton, snom, Grandstream
2010-03-11 - Announcing the 24-hour VoIPathon VoIP chat live to celebrate the three years of VoIP Users Conference
2010-03-11 us philippine voip
DTH Software, Inc releases new Express edition of its billing system for start-ups. 1-24 concurrent calls.
2010-03-10 - Auto-provision function now starts to appear on small embedded IP PBX: MyPBX full series from Yeastar.
2010-03-10 - Apstel Visual Dialplan Voice over IP audit
Asterisk - Run Through with Screenshots.
2010-03-10 - internet telephony voip
Asterisk func device_State
DEVICE_STATE(device)
--Original page content moved from DEVSTATE() wiki page
Synopsis
Get or Set a device state
Introduced in Asterisk 1.6 as DEVICE_STATE(), with a backport available for 1.4 as DEVSTATE().
Description
The DEVICE_STATE function can be used to retrieve the device state from any device state provider.
is the current way to get a devices state.
For example:
NoOp(SIP/mypeer has state ${DEVICE_STATE(SIP/mypeer)})
NoOp(Conference number 1234 has state ${DEVICE_STATE(MeetMe:1234)})
The DEVICE_STATE function can also be used to set custom device state from the dialplan. The "Custom:" prefix must be used.
For example:
Set(DEVICE_STATE(Custom:lamp1)=BUSY)
Set(DEVICE_STATE(Custom:lamp2)=NOT_INUSE)
You can subscribe to the status of a custom device state using a hint in the dialplan:
exten => 1234,hint,Custom:lamp1
The possible values for both uses of this function are:
UNKNOWN | NOT_INUSE | INUSE | BUSY | INVALID | UNAVAILABLE | RINGING | RINGINUSE | ONHOLD
Asterisk 1.6.1.x: The event infrastructure in Asterisk got another big update to help support distributed events. It currently supports distributed device state and distributed Voicemail MWI (Message Waiting Indication). A new module has been merged, res_ais, which facilitates communicating events between servers. It uses the SAForum AIS (Service Availability Forum Application Interface Specification) CLM (Cluster Management) and EVT (Event) services to maintain a cluster of Asterisk servers, and to share events between them. For more information on setting this up, see doc/distributed_devstate.txt.
Troubleshooting
Make sure you know what will happen after an Asterisk restart! It might be necessary to use a .call file (or the Asterisk manager API) to call the DevState application right after Asterisk has started to ensure correct LED status. Note that, before doing so, you might also have to reboot or initialize the phones in question so that they can renew their SIP subscription of the extension that is used to monitor the devicestate; for example SIP NOTIFY could be used for that purpose (see sip_notify.conf).
Example of using DEVICE_STATE for call-limit
Because call-limit is deprecated, sometimes you will need to make sure that, if an extension is in use, you will not call it.
The following dialplan entries make sure that extension 100 has only one call at a time.
exten => 100,1,ExecIf($[ ${DEVICE_STATE(SIP/${EXTEN})} = INUSE ]?Busy)
exten => 100,2,Dial(SIP/${EXTEN})
Example for controlling BLF lights:
See http://www.voip-info. ... 99 per month phone vonage line included minutes
Asterisk Consultants Italy
vonage wav dei consulenti Asterisk in Italia, suddivisi per regione e provincia, free internet phone calls
forniscono consulenze per l'installazione e l'amministrazione di sipser 3.8 c
telefonici basati su Asterisk.
E' possibile avere supporto online in italiano attraverso la rete IRC freenode, canale #asterisk-it
Page ContentsRegione Abruzzo
Provincia Pescara
Regione Free PC to Phone Call No Charge Matera
Regione Calabria
Provincia Reggio Calabria
Regione Campania
Provincia Napoli
Provincia Caserta
Provincia Salerno
Regione Emilia Romagna
Provincia vonage hack
Forli-Cesena
Provincia Rimini
Provincia Modena
Provincia Piacenza
Provincia Reggio Emilia
Regione Friuli Venezia Giulia
Provincia Gorizia
Provincia Trieste
Provincia Udine
Regione Lazio
Provincia Frosinone
Provincia Latina
Provincia Roma
Regione Lombardia
Provincia Bergamo
Provincia Brescia
Provincia Cremona
Provincia Milano
Provincia Pavia
Provincia Varese
Provincia Monza e Brianza
Regione Marche
Provincia Ancona
Provincia Pesaro-Urbino
Regione Piemonte
Provincia Cuneo
Provincia Torino
Regione Puglia
Provincia Bari
Regione Sardenia
Provincia Sassari
Asterisk config skinny.conf
I got this information from Narem. I thought it might help someone.
I will try to configure my Cisco IP Phone 30 VIP with the information below. When I get the 30 VIP phone to work I will post the exact steps and then I will try to get some old Cisco Access Analog Gateway (AS-8) to work by modifiying the skinny.conf file. I will keep you posted. If you have anything to add, please do.
Thanks Narem.
Hi Frank,
I had to download the latest source from CVS and
build. The following link shows how to configure the
phone manually specifying the IP address manually.
Set the Gateway, and TFTP to be the Asterisk server's
IP address.
http://www.cisco.com/documentation/ccn/v22/phone_dhcp_disable.htm
Here is the entry from skinny.conf
Rubbermaid Sipp n Sport
/>device=SEPXXXXXXXXXXX ;; Replace XX with MAC addr.
version=P002F202 ;; This should match the version in the phone/TFTP
context=default
line => 100
I had 3com voip problems
restart Asterisk for successful registration
for the first time. NOTE: This is important. A reload will not work; you must restart!
Naren
vonage voicemail indicator
Configuring Cisco 12SP phones with Asterisk
Configuring Cisco 12SP+/30 VIP phones with Asterisk
This page documents how you configure a Cisco 12SP+/30VIP phone with Asterisk.
The Cisco 12SP+/30VIP IP Phones are EOL and they are not supported by Cisco anymore.
Cisco states that they do not have any listings for this phone and offer no support or downloads for it anymore.
What you need:
A Hex Editor of some kind
TFTP server
Firmware image file ( P002L2J2.bin 128368 bytes )
Let's start off with SEPDefault.cnf.
Reliance VOIP is a 17-byte long vonage security disa
file not an ASCII text file
Open up your SEPDefault.cnf in your hex editor and look for the highlighted section in the example.
The hexadecimal breakdown is here:
Offset 0x0: Header:
01 01 00 01 02
Offset 0x5: Server IP:
C0 A8 01 5A
Offset 0x9: Buffer:
01 03
Offset 0xB: Port number (2000):
D0 07
Offset 0xD: Footer, EOF:
00 00 01 FF
Use a binary calculator such as "bc" to calculate the value.
You can also use the standard calculator that comes with Windows to make these cconversions. Just click on the View menu and switch to Scientific mode. A full binary/hex/octal/decimal calculator is available in this mode.
vonage phone service review bc
bc 1.06
Copyright 1991-1994, 1997, 1998, 2000 Free Software Foundation, Inc.
This is free software with ABSOLUTELY NO WARRANTY.
For details type `warranty'.
obase=16
192
C0
168
A8
1
1
90
5A
quit
bash-2.05b#
In this example, CO A8 01 vonage consumer report
is the IP address of my * server. ...
Cheapest ATAs and Service
Created by Damian with help from Joseph Arsenault - Last Major clean up of list on June 17th,2009
Here's a list of the Cheapest SIP Outbound calls, DID's Per country and ATA Adapters
Page ContentsFree PSTN DIDs & Minutes - This is ONLY for free (as in no catch) PSTN DIDs & Minutes.
Shared DIDs (You call a central number, and put in your Extension, or Enum):
Free Minutes to Various Countries
PSTN DIDs (Ranking: By Country, Provider Name)
Minutes (Tiered rates to US48)
Minutes (Ranking: By Country, Termination Cost)
Unlimited Plans (Ranking: By Country, Provider Name)
ATAs - NO LINKING TO HOME PAGE! ONLY TO PRODUCT PAGE
Phones- NO LINKING TO HOME PAGE! ONLY TO PRODUCT PAGE
Rules
DO NOT remove your competitors unless you are management cost savings for VOIP
than they are (AFTER shipping)
DO NOT post your store's front page, only linking directly to the product mentioned will be allowed (You WILL be deleted for this one)
Microsoft H.323 Telephony
post your prices (listings without prices will be removed without even checking if you're the cheapest)
DO link to your store, but be aware that if you're not the cheapest you might be removed.
DO sort listings with cheapest listing on top (if more than one is provided)
And please add REFURBISHED products as ANOTHER product (see Sipura SPA-2000 example, below).
When comparing prices on hardware, it's wise to check the final price, including shipping voip security vonage .mil
taxes.
Free PSTN DIDs & Minutes - This is ONLY for free (as in no catch) PSTN DIDs & Minutes.
Romania GeoTel Low cost calls and free dial in numbers in Romania.
UK - Localphone Localphone provides free geographic number of most of UK cities just by registering for SIP account. Their service is reliable and call charges are very low.
USA - www.fonosip.com FonoSIP USA DIDs - Free VoIP Account plus DID area code 206 253 360 or 425
USA - www.ipcomms.net FREE USA DIDs - 1 Number + 2 Lines = FREE SIP Delivery
USA - IPKall Free DID's in USA 206, 253, 360, 425 (Seattle/Tacoma WA area)
USA - Sipix SC-1300 digital camera manual
New Software Releases
Page ContentsArchive
March 2010
February 2010
January 2010
This page is to inform on various VoIP related software releases.
Archive
2007
2008
2009
Your contributions are welcome, but please read the How to add information to this wiki page and the Posting Guidelines before you post.
March 2010
2010-03-12 - REMWAVE OS X Communicator as open source - REMWAVE releases source code for SIP softphone for Mac OS X
2010-03-01 - Orgasmatron 5.1 for Asterisk - Free U.S./Canada phone calls plus 30 free VoIP apps (15-minute install)
February 2010
2010-02-24 - SafiServer and SafiWorkshop Version asterisk voip hosted pbx
Beta - Update to the opensource graphical IVR designer and server engine (for Asterisk)
2010-02-22 - Taridium ipbx Taridium releases ipbx eXpress free 5 user edition
2010-02-20 - HostingVOIP release new vippie for voice & SMS VIPPIE for Windows, Symbian, Andriod & Windows Mobile.
2010-02-20 - OfficeSIP Softphone 1.0 is released. Free voice & video softphone for Windows.
2010-02-15 - proyecto VOIP
Software, Inc releases version 5.0 of its VoIP Billing System. Release Notes.
2010-02-12 - UnitePBX. ... vonage tivo
VOIP Service Providers Business Europe
This is a list of VOIP Service Providers who offer full service products primarily aimed at the small to medium sized business telephone market. Such companies typically support multi-line telephone systems, small PBX gateways and hosted VoIP (as an alternative to Centrex service). See also:
VOIP Service Providers Residential Single line residential/business plans go here.
VOIP Service Providers B2B Bulk origination/termination goes here.
RIP VOIP VOIP provider cemeteryâ™
Service providers operating in more then one country are listed under each country. "Operating in a country" means a provider that has billing and support staff located in the country, and offers service subject to the regulations and consumer protection laws of that country.
Marketing is NOT ALLOWED on this page. Please describe services in neutral language and normal fonts. Don't bother listing prices--unless you really plan to return and edit them as things change. If you want to add your company, please read the Posting Guidelines for Promoting Products and Services. When you add your entry to this page, please make sure your entry is in alphabetical order in relationship to other vendors listed in the same section. Failure to follow these guidelines will result in deletion!
Please include relevant information like SIP or IAX handoff, how outgoing CLID is set, whether a given account may originate multiple calls at once, etc.
If you like this page, please link to it, so Google and other search engines will consider it more important.
Users: Please feel free to REMOVE any listing that does not meet the stated goals of this page. ...
VOIP Service Providers Residential
Service Providers Residential
Africa
Logic Ring Very competitive International VoIP Reseller programs for agents and private label resellers.
IPtransit Residential and SME Sip, H323, SIP URI, Inbound, LoCall Plans, PC and Mobile Phone clients, well suited to VSAT, SCPC and contended bandwidth clients, 723.1, &29, 711 and true T38, Calling cards and GSM Gateways, BYOD welcome, test accounts.
Corelynx Inc Hosted Enterprise Telephony for SME and Residential, Full Suite of Call Center Solution - Hosted and Onsite Model, Offers DID and 8XX numbers to more than 50 countries of the world. Capable of offering VOIP based IP PBX Solution on MPLS and VPN for countries where VOIP ports are blocked, IPLC and Colocation Services, A-Z wholesale and retail termination. Also offers any form of customization work on Asterisk or SER platforms.
INDIGICOM VoIP Solutions Provider. Business services, end-users, satellite VoIP.
MyKanKan VoIP Service Provider. Business solutions for call-centers, hotels, callshop .... Voip and Asterisk consulting. Features include : Voicemail, Caller ID w/Name, Call Waiting, Call Forwarding, tenor voip multipath products
ID Block, Free In-Network Calling, Web Based Call Logs. Optional Services include: SoftPhone Access, Virtual Phone Number.
http://www.powerbillbox.com African VoIP carrier and billing provider. African PSTN calling and peering, national & international rates in H323, SIP, IAX.
Takalam Takalam offers VoIP solution starting from $10 to small and medium business in all African and Arab countries like Saudi vonage phone service corp holdings broadband
UAE, Jordan, Egypt, Bahrain, Kuwait, Iraq, Syria, Palestine, Kenya, Nigeria, Rwanda, etc...
TuT Telecom - ABSOLUTELY FREE Peer-to-peer SIP calls vis softphones or SIP hardware. Local DIDs also available to over 7000 cities. Asterisk Supported. PC2Phone services and voip services. WiFi & GSM Voip Phones and services available.
South Africa
Asterisk Directory South Africa Comprehensive Listing of Asterisk Companies and VoIP providers in South Africa
Logic Ring Very competitive International VoIP Reseller programs for agents and private label resellers.
FutureFone South directv and vonage
VOIP provider. SIP based account. ...
Asterisk RealTime PostgreSQL
So, rather than have this stuff all over the VoIP solutions tx I thought it would be easier if I just made a page that would handle all things RealTime and PostgreSQL.
While PostgreSQL can still be used Asterisk cdr pgsql, it can no longer be used with the voicemail system. Thus, if you are using the old pgsql for your db driven voicemail, when you upgrade, you will have to use the unixODBC code. As such, I figured it was just as easy to use the unixODBC subsystem for the CDR. I haven't done any preformance testing yet, but my guess is that the over head is not enough to justify having two different DataAccess layers. Anyhow, I choose to use one, so here is how I got RealTime and the CDR subsystem to use unixODBC.
These are the tables that I used to make my Asterisk RealTime work. I also am including a modified cdr table. I am planning on using the system with an online replicator (http://www.commandprompt.com), which means that all tables must have a primary key. As such, I added the primary key to the cdr table create statement. Lastly, I am put in some reasonable (for us anyway :P) defaults. I also changed the tables to reflect some of the "size" constraints that we run into. We tend to have very long appdata sections for our extensions since we use the app_sql to do things in the database from within the dialplan.
WARNING: Asterisk versions prior to 1.4.15 suffer a vulnerability in res_config_pgsql. If you want Postgres realtime, update immediately to 1.4.15!
vonage voicemail indicator />NOTE: We are using PostgreSQL 8. ...
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