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Asterisk cmd FollowMe
FollowMe Synopsis Find-Me/Follow-Me application
Introduced with Asterisk 1.4, see patch 5574

Description FollowMe(followmeid|options):

This application performs Find-Me/Follow-Me functionality for the caller as defined in the profile matching the <followmeid> parameter in followme.conf. If the specified <followmeid> profile doesn't exist in followme.conf, execution will be returned to the dialplan and call execution will continue at the next priority.
Forked (=simultaneous) dialing of multiple numbers in the same step is supported with this application if you'd like to dial multiple numbers in the same followme step.

Options:
   s    - Playback the incoming status message prior to starting the follow-me step(s)
   a    - Record the caller's name so it can be announced to the callee on each step
   n    - Playback the unreachable status message if we've run out of steps to reach the
          or the callee has elected not to be reachable.

Note that every change in vonage theme song must be activated with a "reload app_followme.so" on the Asterisk CLI.

Important note: The app_followme that's in 1.4 right now do NOT make use of any assets in AstDB as described in the following lines.

The settings in followme.conf allow for an entry that points to the astdb, something like this:

number => family/key
number => family/key

Details Scenario Call comes in, outside caller dials "100" Desk phone for user Joe rings. No answer Joe's house phone rings. Joe's wife picks up and hears a voice "Please press any key to accept a call for extension 100." Joe's wife hangs up. Joe's cell phone rings. Joe picks up and hears a voice "Please press any key to accept a call for extension 100." Joe presses 1 and says "Hello this is Joe".
Alternately, in

H.323 MCU price

penultimate step
Cell voice mail picks up. Voice says "Please press any key to accept a call for extension 100". No keys pressed since it's a voice mail Call is routed to Asterisk voicemail.
The a follow-me number to call is specified in followme.conf. You can specify as many of these numbers as you like. They will be dialed in the order that you specify them in the VONAGE THEME SONG file OR as specified with the order field on the number prompt. Therefore forked (parallel) dialing of multiple numbers in the same step is supported.

Often people want to force their 'follow me' to simply be on or off. So, when it's on, it doesn't ring their desk phone at all, just goes straight to cell/etc. When it's off, it only goes to desk phone, and nowhere else. You can achieve thisvia extensions. ...

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.

internet phones and adaptors calling inphonex account Editing pages to eliminate SPAM and inappropriate content is appreciated, when there are disagreements about what is appropriate content please contact: support@voip-info.org.
If you are removing content from pages, please leave a polite comment explaining the reason for the changes. Uncivil comments, user names, or content will result in the removal of the users editing privileges.

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 - VoIPon Brings the Best of CeBIT to You VoIPon

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Sangoma, AVM, Pika, Patton, snom, Grandstream 2010-03-11 - Announcing the 24-hour VoIPathon VoIP chat live to celebrate Free PC To Phone For USA three years of VoIP Users Conference 2010-03-11 - DTH Software, Inc releases new Express edition of its billing system for start-ups. 1-24 concurrent calls. 2010-03-10 - Apstel Visual Dialplan for Asterisk - Run Through with Screenshots. 2010-03-10 - Phone System Comparison Chart for Spring 2010 Released by CompareBusinessProducts.com - over 90 phone systems compared 2010-03-10 - VoIPon Interviews Rhino Equipment Corp. ... Rubbermaid Sipp n Sport

Asterisk v1.2 upgrade to v1.4 gotchas
The upgrade is fairly smooth, and there's lots of new features. It's the deprecated stuff that bites you during the migration process.

Start the migration with reading UPGRADE.txt, and then look at the CHANGES file for details that were introduced with 1.4.0.

Make sure to read the new xxx.conf.sample files. That way you may detect new features/options that not seldomly also fix potential security issues.

asterisk.conf For sure you will want to have "internal_timing=yes"!

extensions.conf Hurray, you may now monitor the call park and Meet conference with hint, use "Meetme:1234" or "park:701@parkedcalls"!
Call pickup has changed, in particular you really must take a look at PICKUPMARK.

A line starting with ;-- (semicolon immediately followed by two dashes) is now treated as opening a multi-line comment, so be aware! You might disable the entirety of what is remaining in your dialplan from this point on.

Changes to watch out for:
Calling a voicemail box with flags for busy or unavailable (options b and u) must now be performed with a pipe as opposed to prepending that option to the mailbox number: "b1234" or "u4567" turns into "1234|b" and "4567|u" SIP_HEADER() with (Via) now needs to be written as (Via,1) in Asterisk 1.4 LookupCIDName is deprecated. Please use the much more beautiful and easy-to-read Set(CALLERID(name)=${DB(cidname/${CALLERID(num)})}) instead
Many similar changes for variables are described in ugprade.txt:
change ${TIMESTAMP} variable to ${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)} function change ${CALLERIDNUM} variable to ${CALLERID(num)} function

sip.conf In the [general] section "port=" has been renamed to "bindport=" to prevent misunderstandings The default for QoS settings has changed from the old TOS to the new DiffServ method. This also applies to iax.conf, by the way. with the new subscribemwi=yes we can finally instruct Asterisk to not send what some SIP devices consider as unsolicited NOTIFY messages (AVM Fritz!Box, Siemens Gigaset and others). This prevents SIP ERROR 481 or "Remote host cannot match NOTIFY" BLF and voip books you will need to set "call-limit=" to make hints (SIP SUBSCRIPTIONS) work in Asterisk 1.4 also look at the general setting "limitonpeers=yes" and "notifyringing=yes" etc.

iax.conf The new threading model is great, but the default of 10 threads is way too low. Symptoms include total loss of audio until the channel is hung up.

in general section, add: iaxthreadcount = 200 in general section, add: iaxmaxthreadcount = 1000
Later in 1.4.26.2 also this changed due to a security issue:

add this to iax.conf: calltokenoptional = 0.0.0.0/0.0.0.0
add this to the [guest] user in iax.conf: requirecalltoken=no (many guests will be using old Asterisk boxes)
In future: Upgrade the IAX peers and provide call tokens!

zaptel turns into dahdi During the summer 2008 and after the release of 1.4.17 (?) zaptel has been renamed to dahdi. Since zaptel/dahdi provide timing to H.323-compliant Internet telephone software this also matters for users that do not have any zapte (Digium) hardware (ztdummy vs. dahdi_dummy). ...

DTH VoIP Billing and Customer Management
Billing Software for Asterisk
web site http://www.dthvoipbilling.com.com

DTH Billing and Customer Management is a prepaid and postpaid billing application for single-tenant and multi-tenant environments.

We can rate your CDR directly from Asterisk or most any custom source with minor modification. We can bill for DIDs, extensions, trunks, IP addresses and additional recurring charges.

Our system is ideally suited for companies who provide services to residential, wholesale and small business customers including hosted/virtual PBX.

Customers can access their accounts online to review invoicing history, make payments, view CDR and edit billing information.

Additionall information http://www.dthvoipbilling.com

NEWS :
2010-03-13 - DTH announces new Express Edition voip billing software for start-ups. Up to 25 concurrent calls.

CONTACT :

sales@dthsoftware.com

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 vonage commercial 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

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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. ... wireless voip broadband router

Asterisk Consultants Italy
Lista dei consulenti Asterisk in Italia, suddivisi per regione e provincia, che forniscono consulenze per l'installazione e l'amministrazione di centralini telefonici broadband phone service su Asterisk.
E' possibile avere supporto online in italiano attraverso la rete IRC freenode, canale #asterisk-it

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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

Vonage Customer Satisfaction

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

[frankjmvip1]
device=SEPXXXXXXXXXXX ;; Replace XX with MAC addr.
version=P002F202 ;; This should match the version in the phone/TFTP
context=default
vonage snowmobile clip => 100

I had to restart Asterisk for successful registration
for the first time. NOTE: This is important. A reload will not work; you must restart!

Naren

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.

sipser 3.11 solution 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.
Free PC To Phone For USA />SEPDefault.cnf is a 17-byte long BINARY 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

vonage wav advertisement 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.

bash-2.05b# 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 5A is the IP address of my * server. ... Disadvantage of VoIP

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 cheaper 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)
DO post your prices (listings without prices will be removed without even checking if you're the Voice over IP VoIP Netzwerke LAN />DO link to your store, but be aware that if you're internet phone comparisons the cheapest testing skills required for VOIP 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 and taxes.

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USA -

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 insulated sipper BOTTLE to add information to this wiki page and

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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 vonage free activation 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 - DTH Software, Inc releases version 5.0 of its VoIP Billing System. Release Notes. 2010-02-12 - UnitePBX. ... sip codes

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