VOIP-info.org Page Changes

asterisk manager events

Wed, 01 Aug 2007 08:25:03 GMT

User	Comment	Time	Version
chandave	Add Registry event	2007-08-01 08:25	10

Diff from version 9 to 10 (current)

@@ -40,7 +40,21 @@

Uniqueid1: 1094154427.10

Uniqueid2: 1094154427.11

+!!!'Registry' Event

+__Description__:

+# Fired when Asterisk registers with a peer

+__Notes__:

+For an entry like:

+register => username:password:authname@sip.domain:port/local_contact

+''Domain'' would reflect the value of ''sip.domain''

+

+__Data Sample__:

+ Event: Registry

+ Privilege: system,all

+ Channel: SIP

+ Domain: ''sip.domain''

+ Status: Registered

!!Unformatted and Undocumented

Asterisk

Wed, 01 Aug 2007 04:16:42 GMT

User	Time	Version
james.zhu	2007-08-01 04:16	885
james.zhu	2007-08-01 04:15	884
james.zhu	2007-08-01 04:07	883
james.zhu	2007-08-01 04:06	882
james.zhu	2007-08-01 04:02	881
james.zhu	2007-08-01 03:44	880
james.zhu	2007-08-01 03:42	879
james.zhu	2007-08-01 03:29	878
eran	2007-07-31 14:15	877
yeastar	2007-07-31 11:04	876

Diff from version 875 to 885 (current)

@@ -21,11 +21,13 @@

!News

{rss id=2}

+* 2007-08-01 - OpenVox Starts [http://www.openvox.com.cn| an Asterisk Hardware Promotion! ]

+* 2007-07-31 - Xorcom offers [http://www.xorcom.com/documentation/white_papers/fax_modem|Full Fax and Modem Support for Asterisk]

* 2007-07-27 - Come to the First [http://www.danielaliaman.com/maug/|Malaysian Asterisk User Group] Meeting at 7th August at Social Bangsar, (Starts at 19:30)

* 2007-07-24 - [http://www.apstel.com|Visual Dialplan for Asterisk] beta release is available for download - innovative development platform for Asterisk dialplan

* 2007-07-22 - New [http://www.xorcom.com/products/astribank_bri|8-Port BRI ISDN Interface for Asterisk]

* 2007-07-20 - [http://www.xorcom.com/products/xr0030_astribank_6fxs_2fxo|Astribank] new model - 6 FXS + 2 FXO - for small Asterisk implementations

-* 2007-07-20 - [http://www.yeastar.com|New TDM800] No need connector, Red light for FXO and Green light for FXS.

+* 2007-07-20 - [http://www.yeastar.com|New TDM800] No connector need, Red light for FXO port and Green light for FXS port.

* 2007-07-17 - [http://news.yahoo.com/s/ap/20070717/ap_on_hi_te/sunrocket_closes_1|Sun Rocket] suddenly closes its doors

* 2007-07-14 - [http://www.bicomsystems.com|Bicom Systems] releases PBXware 2.0 which includes FoIP and instant messaging.

* 2007-07-14 - ((SiSky|SiSky Enterprise Edition 1.1.0.2)) released - Generate up to 16 Skype Trunks for Asterisk/IPPBX. [http://www.yeastar.com/download/all/SiSkyEEUser.asp|Free Trial].

Yeastar

Wed, 01 Aug 2007 03:50:07 GMT

User	Comment	Time	Version
yeastar		2007-08-01 03:50	25
yeastar		2007-08-01 03:32	24

Diff from version 23 to 25 (current)

@@ -16,7 +16,7 @@

Standard Package: 1 main card; up to 4 daugther modules ; 4 line cables and 1 power cable; Instruction sheet. All are packed in a carton box for safe shipping.

-!{img src=http://www.yeastar.com/images/siskype.gif} Business Skype Solution for Asterisk/IPPBX

+!{img src=http://www.yeastar.com/images/siskyee.gif} Business Skype Solution for Asterisk/IPPBX

Nowadays, Skype is very popular and you may found many customers are Skype users. Let your customers who are used to Skype to contact with you quickly and conveniently is becoming the main job of your Asterisk/IPPBX system. SiSkypeEE is the best solution for you to connect SIP and Skype world.

SiSkypeEE is tailored for Asterisk/IPPBX users and small to medium sized companies that want to bring Skype trunks into the existing system.

@@ -25,7 +25,7 @@

3. Provide one more choice of making landline/mobile calls through SkypeOut

4. Receive calls from website when your customer browsing your webpage

-!{img src=http://www.yeastar.com/images/SiSkypePE.gif} Make/Receive Skype Calls by IP Phone

+!{img src=http://www.yeastar.com/images/SiSkype.gif} Make/Receive Skype Calls by IP Phone

If you own a SIP device, either SIP phone or SIP ATA is OK, which connect to your company's IPPBX or ITSP's SIP server, and sometimes you want to get cheaper and better calls through Skype or your customers is only available on Skype, in this case you must use earphone or USB phone to make or receive Skype calls. With SiSkype Personal Edition, an IP phone is enough to meet your demand. A neatly tabletop will do a greatly convenience to your work.

SiSkypePE is tailored for those users who own a SIP device that connect to their company's IPPBX or ITSP's SIP server to make and receive Skype calls on their existing SIP device.

@@ -35,17 +35,54 @@

4. Transfer Skype Calls to other IP phones

5. Quick and easy operation and excellent Voice Quality

-!!__[http://www.yeastar.com/ProductsforPBX.asp|Product for PBX User]__

-!{img src=http://www.yeastar.com/images/xskype.gif} Business Skype Solution for PBX

-XSkype is a business class Skype gateway with 16 ports. Bringing skype into SMBs existing PBX, XSkype allows them to make and receive calls of Skype and PSTN over the same office phone in the comfortable way that they have got used to.

-Saving the outstanding features of Skype flexibly and extending all the activities from PC to the extensions of the whole business, XSkype enables SMBs to enjoy the domestic and international calls freely and cheaply in crystal clear voice.

-Using XSkype, SMBs can take full advantage of the largest free VoIP in the world to communicate and collaborate with clients and partners easily, efficiently and economically.

-XSkype is tailored for small and medium sized businesses (SMBs). It is one of the best solutions for those companies who want to upgrade their existing PBX to VoIP (Skype) and those that with multiple offices on different locations. It also provides WebCall to let customers make calls to your company from website.

+!!__[http://www.yeastar.com/ProductsforPBX.asp|Product for Traditional PBX User]__

+!{img src=http://www.yeastar.com/images/exsky.gif} Business Skype Solution for Traditional PBX

+ExSky is a business class Skype gateway with 16 ports. Bringing skype into SMBs existing PBX, ExSky allows them to make and receive calls of Skype and PSTN over the same office phone in the comfortable way that they have got used to.

+Saving the outstanding features of Skype flexibly and extending all the activities from PC to the extensions of the whole business, ExSky enables SMBs to enjoy the domestic and international calls freely and cheaply in crystal clear voice.

+Using ExSky, SMBs can take full advantage of the largest free VoIP in the world to communicate and collaborate with clients and partners easily, efficiently and economically.

+ExSky is tailored for small and medium sized businesses (SMBs). It is one of the best solutions for those companies who want to upgrade their existing PBX to VoIP (Skype) and those that with multiple offices on different locations. It also provides WebCall to let customers make calls to your company from website.

1. Place SkypeOut calls from your existing office phone

2. Connect multiple branches easily and freely

3. Receive calls from website

4. Receive SkypeIn Calls over your existing office phone

+

+!{img src=http://www.yeastar.com/images/bizsky.gif} Call Manager & Romote Extension for Traditional PBX

+BizSky is hardware-software integrated design solution that runs on Windows OS applied to traditional PBX system, in short, BizSky is an advanced business calls manage system. Every company has unique services and different requirements from diversity of operating posts, with the Yeastrar's BizSky solution, all posts in SMBs can have their communication and networking needs met flexibly.

+To every company, especially to every advanced phone users, the value of a call or contact information is vital to competing in global economy. BizSky solution that provides customized call routing, live and voice mail and recording backup, caller ID visible, click to SkypOut easily or click to transfer incoming calls throughout the enterprise as well as PSTN networking to enhance effective team operations, which are useful and powerful connectivity in attracting valuable customers.

+Nowadays, many advanced phone users are busy in their frequent business trips. Providing them both seamless communications and access to advanced productivity tools without wasteful duplication of investments is the key to survival and growth. BizSky provides unique capabilities to make Skype works as an extension phone when advanced users working in a remote office.

+!!Functions:

+Call Manager for Advanced Telephone Users:

+1. Remote Office: Use Skype as office extension when working in remote office.

+2. Mobile Office: Forward office calls to mobile phone while you are away.

+3. Voice Message: Let caller leave voice massage when you are busy or away.

+4. Call Record: Click to record important call.

+5. Caller ID Profile: Show caller's information in real-time.

+6. Click to Call: Click on 'name' to call, don't waste your precious time on dialing.

+7. Conference Call: Add related person into conference, save communication time.

+8. Call History: Easy to review log, no more missed business calls!

+Skype Trunks for Every One:

+1. SkypeOut: Place SkypeOut calls from existing office phone.

+2. United Office: Connect multiple branches easily and freely.

+3. Web Call: Publish company unique Skype ID, and receive business calls from website visitors through multi-trunk.

+4. Virtual Office: Apply local SkypeIn telephone numbers in different countries and it's a local call for customers in those countries to reach you on regular phone.

+!!Features:

+1. PSTN-FALLBACK: If BizSky server down, you still can make/receive PSTN calls

+2. TRANSPARENCY: Keep all PBX functions and habits for advanced phone users.

+3. SAVE-MONEY: Make SkypeOut call from any extension.

+4. WEB-CALL: Website visitors click to call your company.

+5. REMOTE-OFFICE: Work at home or hotel, use Skype as extension phone.

+6. VISUAL: Monitor and Control calls by BizSky Client.

+Other Features:

+1. Click to transfer PSTN or VoIP (Skype) call.

+2. Extension list, click to call or transfer.

+3. Contacts list, edit and click to call.

+4. Share contacts between advanced phone users.

+5. Follow-me, you can make/receive office call anywhere.

+6. Speed dial, Easy to make Skype2Skype call and keep dial habit.

+7. Direct line, Easy and quick to connect branch offices.

+8. Custom Automatic Attendant and Voice Message Greeting.

+9. Sound files save as standard WAV file format.

For more information:

[http://www.yeastar.com|http://www.yeastar.com]

Windows

Wed, 01 Aug 2007 01:29:32 GMT

User	Comment	Time	Version
yeastar		2007-08-01 01:29	13

Diff from version 12 to 13 (current)

@@ -13,4 +13,4 @@

* (asterisk + chan_celliax + related configuration utilities) runs on Asterisk 1.2 on Linux and Windows (with cygwin and a little modification to the compilation of Asterisk to avoid the calls to sigkill). Supports IAX, SIP, and cellphones (with support for Skype calls from/to cellphones). With Celliax is distributed the ((Celliax LiveCD)), with a working installation of Asterisk, chan_celliax, and configuration utilities, based on Knoppix. The Celliax LiveCD contains also all is needed to run Celliax on Windows: the cygwin installer and the tgz with the asterisk-celliax stuff to be untarred in a basic cygwin installation.

!!![http://www.yeastar.com|Yeastar]

-* TDM800 analog interface card that support windows2000/xp/2003

+* TDM800 analog interface card that support windows2000/xp/2003/Vista

voip-info.org

Wed, 01 Aug 2007 01:21:36 GMT

User	Comment	Time	Version
ZX81	VentureVoIP releases AgentPopper	2007-08-01 01:21	7860
vwarevoip		2007-07-31 20:22	7859
eran		2007-07-31 14:13	7858

Diff from version 7857 to 7860 (current)

@@ -8,6 +8,9 @@

#################################################

##### INSERT NEW STORIES JUST AFTER THIS LINE############{/COMMENT}

+* 2007-08-01 - [http://www.venturevoip.com/news.php?rssid=1802|VentureVoIP releases AgentPopper] - free software for screenpopping with Asterisk 1.4 and Agents

+* 2007-07-31 - [http://www.myvoipspeed.com|MySpeed VoIP] v7 tests connections for VoIP Quality - [http://www.visualware.com/company/pressroom/press/pr20070731.html|press release]

+* 2007-07-31 - Xorcom offers [http://www.xorcom.com/documentation/white_papers/fax_modem|Full Fax and Modem Support for Asterisk]

* 2007-07-30 - [http://msg1svc.net/servlet/FormListener?Y2JleW9uZDoxOTpMTg==|Denver Cbeyond SIPconnect Forum 08/02/2007] FREE event, an industry update on SIPconnect by CTO Chris Gatch

* 2007-07-30 - [http://www.switchvox.com/sv?page=press_room/switchvox3|Switchvox] releases version 3.0, [http://www.tmcnet.com/channels/open-source-crm/articles/9020-switchvox-smb-v30-now-available-support-real-time.htm|touts Salesforce and SugarCRM integration].

* 2007-07-27 - [http://www.astawerks.com/newsdesk_info.php?newsdesk_id=28|AASTRA Releases new Firmware]AASTRA Releases Version 2.1 of Firmware for 5i SIP Phones

@@ -25,6 +28,7 @@

* ((New Software Releases|Software Releases)) - Check here for recent VoIP-related software releases.

* ((new VOIP services|VoIP Services)) - Check here for news on Voip Services.

* ((Old News|Older News)) - Check out older news articles here

+

!! Getting Started

Asterisk cmd Set

Tue, 31 Jul 2007 22:08:31 GMT

User	Comment	Time	Version
johnlange		2007-07-31 22:08	14

Diff from version 13 to 14 (current)

@@ -23,6 +23,7 @@

Set(numTries=4)

Set(CALLERID(number)=000000)

Set(CALLERID(name)="The Name")

+ Set(NIGHTMODE=1,g) ; set a global variable

__ NOTES: __

* Variable names are not case sensitive.

Asterisk cmd Dial

Tue, 31 Jul 2007 21:02:50 GMT

User	Comment	Time	Version
johnlange		2007-07-31 21:02	132

Diff from version 131 to 132 (current)

@@ -35,6 +35,7 @@

** __M(__''x''__)__: Executes the macro (x) upon connect of the call (i.e. when the called party answers)

** __h__: Allow the callee to hang up by dialing __*__

** __H__: Allow the caller to hang up by dialing __*__

+** __i__: Asterisk will ignore any forwarding requests it may receive on this dial attempt. (new in 1.4) Useful if you are ringing a group of people and one person has set their phone to forwarded direct to voicemail on their cell or something which normally prevents any of the other phones from ringing.

** __C__: Reset the CDR (Call Detail Record) for this call. This is like using the ((Asterisk cmd NoCDR|NoCDR)) command

** __p__: This option enables screening mode. This is basically Privacy mode without memory of how to handle the caller. It looks for the file ~np~sounds/priv-callerintros/${IF($[ "${CALLERID(num)}" != "" ]?${CALLERID(num)}:NOCALLERID_${EXTEN}${CUT(CHANNEL,/,1)}=${CUT(CHANNEL,/,2)})}.gsm~/np~ and if it is not found, prompts the caller to say his name. It then rings the called party and plays sounds/priv-callpending, sounds/priv-callerintros/<see-above>, and sounds/screen-callee-options. If the called party enters 1, the call is accepted, 2, the DIAL command exits with ${DIALSTATUS} set to NOANSWER, 3, set to TORTURE and 4, set to DONTCALL. If no valid entry is made, the DIAL command exits with ${DIALSTATUS} set to ANSWER. ~~darkblue: The check for pre-existence of the name recording may not be what you want. For example, everyone from the same number is not necessarily the same person, especially if the number is OUTOFAREA, but if the file is there, no new name will be recorded. Since the files are never removed, you may wish to remove them with a ~np~System(rm /var/lib/asterisk/sounds/priv-callerintros/${IF($[ "${CALLERID(num)}" != "" ]?${CALLERID(num)}:NOCALLERID_${EXTEN}${CUT(CHANNEL,/,1)}=${CUT(CHANNEL,/,2)})}.*)~/np~ right before the Dial command and clean up old ones with a cron job. ~~

** __P(__''x''__)__: Use the ((Asterisk cmd PrivacyManager|PrivacyManager)), using ''x'' as the database (''x'' is optional)

VoIP Gateways

Tue, 31 Jul 2007 21:01:12 GMT

User	Comment	Time	Version
philipnguyen	VOXTREAM - Asterisk peering box	2007-07-31 21:01	345

Diff from version 344 to 345 (current)

GXP-2000

Tue, 31 Jul 2007 20:53:15 GMT

User	Comment	Time	Version
easycall		2007-07-31 20:53	1478

Diff from version 1477 to 1478 (current)

@@ -1245,6 +1245,7 @@

* Norway - [http://www.novaoffice.no/|NovaOffice]

* Norway - [http://www.voipbutikken.no/|Voipbutikken.no]

* Norway - [http://televoip.no/|TeleVoIP]

+* Poland - [http://www.easycall.pl/akcesoria_szczegoly.html,di--11,Telefon-IP-Grandstream-GXP-2000|easyCALL]

* Portugal - [http://www.voxsys.com.pt|Voxsys, RSCE] (Distribuidor GrandStream)

* Slovenia - [http://www.tend.si|ORG. TEND ?? evolucija informacijskih sistemov]

* Spain - [http://www.avanzada7.com|Avanzada 7]

Asterisk SIP channels

Tue, 31 Jul 2007 20:31:55 GMT

User	Comment	Time	Version
flobi	adding alternative dialing example	2007-07-31 20:31	108

Diff from version 107 to 108 (current)

@@ -42,9 +42,11 @@

Here are some examples of complete Dial commands as they might appear in your ((Asterisk config extensions.conf|Dialplan)):

exten => s,1,Dial(SIP/ipphone) ; Call our peer "ipphone" whose connection details are in sip.conf

+ exten => s,1,Dial(SIP/joshua@ipphone) ; Call our peer "ipphone", requesting extension "joshua"

exten => s,1,Dial(SIP/john@foo.com) ; Connect to foo.com, requesting extension "john"

exten => s,1,Dial(SIP/192.168.1.8:9999,20); Connect to 192.168.1.8 on port 9999, with a 20 sec timeout.

exten => s,1,Dial(SIP/8500@sip.com:9876) ; Connect to sip.com port 9876, requesting extension 8500.

+

!!Distinctive Ring Styles

There doesn't yet seem to be a standard for how to tell a SIP phone that you want it to ring with a distinctive ring. On SIP handsets that support distinctive ring at all, the exact method of specifying distinctive ring varies from one model to another. Often (or always?) it is by sending a SIP "Alert-info" header, but what the value of this header should be is not consistent. If you can figure out what Alert-info header Asterisk should send, then you can get Asterisk 1.0 and 1.2 to send such a header by setting the ALERT_INFO channel variable before you Dial:

VoIP Security Consultants

Tue, 31 Jul 2007 19:18:30 GMT

User	Comment	Time	Version
zeedo		2007-07-31 19:18	3

Diff from version 2 to 3 (current)

@@ -10,6 +10,8 @@

!!VoIP Security Consultants

Companies that provide security consultant services:

+

+* [http://www.ngssoftware.com|NGSSoftware] Security consultants with expertise in voice security. NGS provide a variety of security services such as infrastructure assessments, source code reviews, zero knowledge penetration tests and application assessments. NGS also conduct research and publish papers and presentations on security issues.

* [http://www.tacvoip.com|Tactical VoIP] Independent security consultants offering contract VoIP security audit, secure design, and forensic investigation services. Available World-wide. Currently serves Fortune 500, Government, and Industrial clients.

Open Source Billing Systems

Tue, 31 Jul 2007 18:51:14 GMT

User	Comment	Time	Version
dlouhan	rollback because of the undesirable advertisement	2007-07-31 18:51	90

Diff from version 89 to 90 (current)

@@ -1,50 +1,3 @@

-!!!AAAVEER ( BILLCALL-COMPLETE CARRIER GRADE BILLING AND RESOURCE PLANNING REAL TIME BILLING SYSTEM )

-

-Projects Handled:

-

-*OPEN SER

-* Asterisk Billing

-*AST2BILL

-*OPEN ASR

-*OPEN LCR

-*VICIDIAL

-*YATE

-*TRIXBOX

-*IPPBX

-*VIDEO CONFERENCING

-

-Services Offered:

-

-* Installation/maintenance/configuration of

--Digium FXO/E1/T1/TDM Cards,

--Soft Switch for Carriers

--Video Conferencing IPPBX with Media Server,

--Call Center Solution

--Broadcasting System

--Class 5 Soft Switch

--Class 4 Soft Switch

--Carrier Billing System

--SER

-

-* Turnkey Asterisk solutions & Integration of asterisk with anyother VoIP architecture like

-

--Audio Codes

--VOIP ATA SIP GATEWAY,

--Radius Design,

--AGI SCRIPTING,

--CALLING CARD MANAGEMENT,

--GATEKEEPER BILLING SOLUTION.

--DID/TOLLFREE NUMB/CODEC MAPPING & INSTALATION.

--INBOUND/OUTBOUND/DOMESTIC CALL CENTER SOLUTIONS

-

-CALL TODAY A COMPLETE ASTERISK KPO SUPPORT AND MAINTAINENCE

-

-!!! Phone: +61-290-372-182(Australia),+254-725-202-503(Africa),+91 9825500318(India), +1-818-804-2994 (USA).

-!!! Email: mahavirds@panamaxil.com

-

------------------------------------

-

!!Asterisk2Billing (A2Billling) :

* [http://www.asterisk2billing.org/]

Asterisk system vendors

Tue, 31 Jul 2007 18:37:29 GMT

User	Time	Version
Jezzv8	2007-07-31 18:37	713
Jezzv8	2007-07-31 18:33	712
Jezzv8	2007-07-31 18:32	711
Jezzv8	2007-07-31 18:30	710
Jezzv8	2007-07-31 18:14	709

Diff from version 708 to 713 (current)

@@ -24,6 +24,7 @@

* [http://www.enterux.com | Enterux Solutions, Digium Hardware, IAXy, Full PBX System Vendor , End to End Asterisk Solution Provider(India)]

* [http://www.yestalk.net | E-Resource 21 Pte Ltd. (Singapore) Digium, IP Phones, ATAs, /Professional service for quintum, cisco, SER, Asterisk]

* [http://www.intelegentnetworks.com | Intelegent Networks (US/India) Asterisk consulting, Converged communication solution]

+* [http://www.ispi-cti.com | Intelligent Solutions Phils., Inc. (Philippines)] - System Integration, Intelligent PBX(PSTN+GSM+VOIP) Solutions, Call Center Solutions, Network Infrastructure Consulting and Solutions.

* [http://www.intellectit.com.au/global/services/voice_over_ip.html | INTELLECT Information Technology Pty Ltd, (Australia)]

* [http://www.iipc.com.au/ | Intelligent IP Communications Pty Ltd, (Perth, Western Australia)]

* [http://www.invitesse.com | The InVitesse Group Ltd (Japan)]

@@ -63,6 +64,7 @@

*[http://www.iphonex.com.my| Netregy Systems Sdn Bhd (Malaysia) IP-PBX Implementation Service and Digium reseller in Malaysia

* [http://www.symphonysolutions.biz | Symphony Solutions (Mumbai, India)]

+

!!Europe

Asterisk Canadian Users

Tue, 31 Jul 2007 17:57:32 GMT

User	Comment	Time	Version
breid		2007-07-31 17:57	151

Diff from version 150 to 151 (current)

@@ -82,6 +82,9 @@

* Toronto: The [http://taug.ca|Toronto Asterisk User Group] (TAUG) meets the last Wednesday of every month at various locations, and welcomes anyone from anywhere who's interested in Asterisk.

!!!Quebec

* Montreal: [http://www.amug.ca/|Montreal Asterisk User Group]

+!!!Winnipeg

+* An Asterisk User Group meets on the first Tuesday of the month at 7:00 PM. If you are interested please join our [http://www.muug.mb.ca/mailman/listinfo/asterisk|mailing list].

!!Mailing Lists

* [http://taug.ca/index.php?page=discuss|Ontario Asterisk and VoIP Mailing List] (merged with TAUG) Send mail to asterisk-subscribe at uc dot org to join.

+* [http://www.muug.mb.ca/mailman/listinfo/asterisk|Winnipeg Asterisk User Group]

VOIP Service Providers Residential

Tue, 31 Jul 2007 17:32:59 GMT

User	Comment	Time	Version
intralanman		2007-07-31 17:32	952

Diff from version 951 to 952 (current)

@@ -728,6 +728,7 @@

* [http://www.zingotel.com|ZingoTel] Serving US, Canada, Latin America.

* [http://www.vonworldwide.com/|vonworldwide.com] Vonworldwide provides SIP and IAX termination (877) 866-5678 or businessdevelopment@vonworldwide.com

* [http://www.whichvoip.com|WhichVoIP.com] Compare and review VoIP Service Providers

+* ((Zingo Telecom)) Zingo Telecom is a leading global broadband telecommunications provider, serving residential and small business users throughout the US, Canada, Latin America, and beyond. (Special deals for former SunRocket Customers)

__Bahamas__

* [http://www.ideasip.com|IdeaSIP] SIP based open network. Free in-network calling. Prepay IdeasIN DIDs and IdeasOUT termination

VOIP Consultants

Tue, 31 Jul 2007 17:22:30 GMT

User	Comment	Time	Version
willamowius		2007-07-31 17:22	823
mpls-ejo	Added Sound Choice Communications LLC	2007-07-31 14:32	822

Diff from version 821 to 823 (current)

@@ -2060,6 +2060,16 @@

Asia/Pacific: rodney@sineapps.com

---

+!!~~#ff0000:Sound~~ ~~#0000ff:Choice~~ Communications LLC

+*VoIP network design services and consulting

+*VICI predictive dialer installations

+*TrixBox(tm) module design

+*TrixBox(tm) support

+

+*[http://www.soundchoicecomm.com]

+*+1.651-999-0888 voice line

+

+---

!!!VegaSystems IT Consulting & Solutions Paderborn / NRW / Germany

@@ -2533,7 +2543,7 @@

---

!!Jan Willamowius

-Jan Willamowius, VoIP Consultant and GNU Gatekeeper project leader

+Jan Willamowius, VoIP Consultant and GNU Gatekeeper (GnuGk) project leader

* VoIP consulting

* GNU Gatekeeper support

@@ -2569,6 +2579,7 @@

VOIP : 754435050@telcomserv.net

---

+

!!x9media ip communication

H.323

Tue, 31 Jul 2007 17:20:12 GMT

User	Comment	Time	Version
willamowius		2007-07-31 17:20	34

Diff from version 33 to 34 (current)

@@ -9,7 +9,7 @@

* ((ATcom)):H.323 to ISDN Gateway

* ((Open H.323)): Open Source implementation

* [http://www.gnugk.org/|OpenH323 Gatekeeper]: The GNU Open Source H.323 Gatekeeper

-* [http://www.telos.de/linux/H323/|ISDN2H323]: H323 to ISDN Gateway

+* [http://www.telos.de/linux/H323/|ISDN2H323]: H323 to ISDN Gateway (discontinued)

* [http://www.gnugk.org/h323-isdn-gw.html|IsdnGw]: H.323 to ISDN Gateway

* [http://www.obj-sys.com/open/index.shtml|ooh323c]: An Open Source C implementation of H.323 stack

* [http://www.uniqall.com/host_media_processing/gridborg_hmp.php|Uniqall Gridborg HMP] Proprietary Host Media processing server with H.323 and SIP frontends, and simple ASCII control protocol. It works in both Linux & Windows environments. Its client-server architecture enables you to use any programming or scripting language. It can handle 240 ports on dual processor servers.

Zingo Telecom

Tue, 31 Jul 2007 17:03:02 GMT

User	Comment	Time	Version
intralanman	A leading global broadband telecommunications provider.	2007-07-31 17:03	1

Diff from version 1 to 1 (current)

@@ -1 +1,44 @@

+Zingo Telecom is a leading global broadband telecommunications provider, serving residential and small business users throughout the US, Canada, Latin America, and beyond. Our next generation voice-over-IP network allows anyone with a touch-tone phone and a broadband connection to call anywhere in the world. We provide unlimited nationwide calling plans, super-low international rates, and full suite of innovative calling and account management features at an unbeatable value. With our cutting-edge business services, any small business can enjoy advanced PBX functionality and multi-line savings that have traditionally been limited to large enterprise systems. Zingo Telecom strives to be the provider of choice for subscribers migrating the next generation of IP communication solutions. Browse our learning area for ways to benefit and save with Zingo Telecom.

+

+[http://www.zingotel.com/online/en/SunRocket|Special Deal To Former SunRocket Customers]

+[http://www.zingotel.com/online/en/|Zingo Telecom Main Page]

+[http://www.zingotel.com/online/en/contact.php|Zingo Telecom Contact Page]

+

+!Configuration Samples

+

+* ((Asterisk))

+^

+~np~[ZingoTel] ~/np~

+ type=friend

+ host=sip1.whistlertel.com

+ port=5060

+ username=PROVIDED_DID

+ fromdomain=sip1.whistlertel.com

+ secret=TOP_SECRET_PASSWORD

+ insecure=very

+^

+

+* ((CallWeaver))

+^

+~np~[ZingoTel] ~/np~

+ type=friend

+ host=sip1.whistlertel.com

+ port=5060

+ username=PROVIDED_DID

+ fromdomain=sip1.whistlertel.com

+ secret=TOP_SECRET_PASSWORD

+ insecure=very

+^

+

+* ((FreeSwich))

+^

+~np~ <gateway name="ZingoTel">~/np~

+~np~ <param name="username" value="PROVIDED_DID"/>~/np~

+~np~ <param name="password" value="TOP_SECRET_PASSWORD"/>~/np~

+~np~ <param name="expire-seconds" value="600"/>~/np~

+~np~ ~/np~

+~np~ <param name="proxy" value="sip1.whistlertel.com"/>~/np~

+~np~ </gateway>~/np~

+^

Cheapest ATAs and Service

Tue, 31 Jul 2007 15:08:11 GMT

User	Comment	Time	Version
echamberlain	Frank don't break my links	2007-07-31 15:08	754

Diff from version 753 to 754 (current)

@@ -896,7 +896,7 @@

[http://www.voipipbx.com/IP-Phones/Linksys-SPA962/Detailed-product-flyer.html|voipipbx.com] $249.95 Ship International

[http://www.voicemeup.com/scripts/logs_hits.php?advertiser=15&destination=home&lang=en | VoiceMeUp] Linksys SPA962 - IP Phone $269.95

-[http://store.voxilla.com/|Voxilla Store] - [http://store.voxilla.com/voip-products/linksys-spa962-voip-phone.html|Linksys SPA962] for $279.95 -

+[http://store.voxilla.com/|Voxilla Store] - [http://store.voxilla.com/voip-products/linksys-spa962-voip-phone.html|Linksys SPA962] for $279.95 - [http://voxilla.com/voxilla-reviews/telephones/linksys-spa962-785-99.html|SPA962 Product Review]

[http://www.voipsupply.com/advanced_search_result.php?keywords=spa+962&x=0&y=0?ref=voipinfo| VoIP Supply] $299.95

[http://www.astawerks.com/product_info.php?cPath=71&products_id=228|Astawerks] - $282.25

[http://voip-stock.com/product_info.php?manufacturers_id=14&products_id=327|VoIP-Stock] $309.00

DigiumCards.com

Tue, 31 Jul 2007 12:58:46 GMT

User	Comment	Time	Version
e42006	Updated Open Box items	2007-07-31 12:58	181

Diff from version 180 to 181 (current)

@@ -1,10 +1,8 @@

OPEN BOX ITEMS-

-(10) IP550

(2) IP301

-(3) IP401

(1) Mediatrix 2 Port FXS Gate

-(9) Linksys 942

+(3) Linksys 942

email -info@e4strategies.com for the complete list

Predictive dialer

Tue, 31 Jul 2007 12:44:01 GMT

User	Comment	Time	Version
pgaz	added predictive simulation model	2007-07-31 12:44	95

Diff from version 94 to 95 (current)

@@ -53,6 +53,7 @@

----

__DIALER AND CALL CENTER REPORTING TOOLS__

* ((QueueMetrics)) can be used to measure an outbound campaign using a dialer.

+* [http://www.call-center-tech.com/predictive-dialer-simulation-model.htm|Predictive Dialer Simulation Model] - measures performance of predictive dialing campaigns including cost and profit model.

----

* ((Asterisk)): Wiki home page | ((Asterisk FAQ|FAQ)) | ((Asterisk introduction|Introduction))

IVR System Simulation from DSC

Tue, 31 Jul 2007 12:18:49 GMT

User	Comment	Time	Version
pgaz		2007-07-31 12:18	6

Diff from version 5 to 6 (current)

@@ -8,9 +8,11 @@

To assist customers and prospects with sizing an IVR system and for estimating agent resources for their particular calling campaign, DSC offers a __FREE__ [http://www.call-center-tech.com/ivr-simulation-model.htm|IVR Software Simulation] service that accepts input parameters and calculates the resources required to perform a given inbound calling campaign. The results project campaign resource utilization and efficiencies.

!IVR and VoiceXML Database Access

-The DSC IVR systems can access information maintained anywhere on the internet using VoiceXML. When estimating average IVR resources, the simulation model must take into account the time required to gather this information and convert it and present this data to the caller. This is sometimes difficult because of the complexity of each request and the availability of the media server that is providing this information.

+The DSC IVR systems can access information maintained anywhere on the internet using [http://www.voip-info.org/wiki/view/VoiceXML+from+DSC|VoiceXML]. When estimating average IVR resources, the simulation model must take into account the time required to gather this information and convert it and present this data to the caller. This is sometimes difficult because of the complexity of each request and the availability of the media server that is providing this information.

+

+!::Simulating Interactive Voice Response Campaigns::

+---

-!Simulating Interactive Voice Response Campaigns

The DSC [http://www.voip-info.org/wiki/view/Interactive+Voice+Response+from+DSC|Interactive Voice Response] model is a resource calculator that estimates the amount of phone lines and supporting telecommunication equipment required to handle different types of automatic phone answering campaigns.

There are two types of IVR campaigns that can be simulated using this model. The first is a simple IVR with menu options that does not provide the caller with an option to speak with an online agent. The IVR administrator only needs to provide an estimate of the average time spent in the IVR by a caller. A more complex call is a __Press Through__ campaign where the caller selects an option to speak with an agent. This requires additional phone resources (if an outside line is required) and a more complex time estimation based upon the conversation time between the caller and the agent.

@@ -36,13 +38,17 @@

* __Telecom Resource Requirements__

* __Agent Utilization__

-For a thorough description of the IVR parameters and to view a sample simulation run including a final display, please visit the [http://www.call-center-tech.com/ivr-simulation-model.htm|IVR Simulation] webpage.

+For a complete definition of the IVR parameters and to view a sample simulation run including the final display report, please visit the [http://www.call-center-tech.com/ivr-simulation-model.htm|IVR Simulation] webpage.

+

+---

!IVR Applications

[http://www.call-center-tech.com/phone-applications.htm|IVR applications] and [http://www.voip-info.org/wiki/view/Voice+broadcasting+from+DSC|Voice broadcasting] applications include emergency alerts, school and church notifications, and commercial customer and prospect contacts. The DSC [http://www.databasesystemscorp.com/psivr.htm|IVR System] can provide information to clients 24 x 7.

*[http://www.voip-info.org/wiki/view/Telephone+reassurance|RUOK] Phone System - calls seniors and elderly at home ensuring they are OK ([http://www.callingcare.com/ruok.htm|RUOK]).

* [http://www.voip-info.org/wiki/view/Emergency+Notification+from+DSC|Emergency Notification Systems] - Community alert systems and [http://www.emergency-broadcasting.com|Emergency Voice Broadcasting] services.

+* [http://www.voip-info.org/wiki/view/Phone+Surveys+from+DSC|Phone Surveys] - IVR [http://www.databasesystemscorp.com/pssurvey.htm|phone surveys] and applications development tools

+* [http://www.voip-info.org/wiki/view/Phone+Reminders+from+DSC|Phone Reminders] - Send phone [http://www.call-center-tech.com/reminders.htm|reminders] to clients and subscribers

!Additional Simulation Models

* [http://www.voip-info.org/wiki/view/Voice+Broadcast+Simulation+from+DSC|Voice Broadcast] Simulation and modeling application

Asterisk hardware

Tue, 31 Jul 2007 09:31:17 GMT

User	Comment	Time	Version
yann.buannic		2007-07-31 09:31	254

Diff from version 253 to 254 (current)

@@ -234,7 +234,7 @@

While Asterisk will run on many different systems, the limitations will be determined by what type of interface cards you choose for connecting to phone lines or phones.

-__PCI Slot Requirements__: Carefully check what the requirements are for any PCI card you select in terms of PCI slot type, 5 volt vs. 3.3V, 32 bit vs. 64 bit, 33 Mhz or higher. The ((X100P clone)) cards require 5 volts and won't work iin motherboards that can't supply 5 volts. The cards should be keyed so they will only fit in 5V slots, but at least some of the clone cards have the notch in the PCI connector that will allow them to fit into 3.3V slots. Symptoms of putting a 5V card in a 3.3V only slot is that the system won't power up.

+__PCI Slot Requirements__: Carefully check what the requirements are for any PCI card you select in terms of PCI slot type, 5 volt vs. 3.3V, 32 bit vs. 64 bit, 33 Mhz or higher. The ((X100P clone)) cards require 5 volts and won't work in motherboards that can't supply 5 volts. The cards should be keyed so they will only fit in 5V slots, but at least some of the clone cards have the notch in the PCI connector that will allow them to fit into 3.3V slots. Symptoms of putting a 5V card in a 3.3V only slot is that the system won't power up.

Pictorial guides to identifying types of PCI slots:

* [http://www.digium.com/en/docs/misc/pci_slot.php|From Digium]

@@ -304,6 +304,7 @@

**[http://usa.asus.com/products/mb/socket478/p4s800-mx/overview.htm|Asus P4S800-MX (SiS661FX,SiS963L Chipsets)]

***This was tested with Asterisk@Home 1.3. All system hardware required was detected (including onboard LAN). There was no problem using onboard video. System was used with a Digium X100P Clone FXO card.

---

+