Hanlong Technology

hanlongtekvoip — Thu, 16 Oct 2008 03:51:40 +0100

Hanlong Technology CO.,Ltd.

Hanlong is a manufacturer of low cost SIP VOIP products.

Products

Unicorn4102 VOIP Phone
Unicorn2101: SIP ATA with Router + FXS port + FXO fallback port (not a REAL FXO port, not SIP Accessible)
Unicorn2002: SIP ATA with Router + two FXS ports
Unicorn2112: SIP ATA with Router + FXS port + FXO port
Unicorn6040: Analog FXO IP Gateway: 4 Port FXO Gateway
Unicorn6080: Analog FXO IP Gateway: 8 Port FXO Gateway
Unicorn6004: Analog FXS IP Gateway: 4 Port FXS Gateway
Unicorn6008: Analog FXS IP Gateway: 8 Port FXS Gateway
Xanadu@home: IP-PBX with one 100M network interface

voip-info.org

cosmicwombat — Thu, 16 Oct 2008 02:07:01 +0100

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.

Your contributions are welcome, but please read the How to add information to this wiki page and the Posting Guidelines before you post.

NEWS

2008-10-16 - Interview-Philippe Lindheimer talks about AsteriskNow 1.5 w/FreePBX
2008-10-15 - TRex - Asterisk- based Carrier Grade Softswitch with up to 40 E1 per chassis
2008-10-15 - Brekeke Software Receives a 2008 INTERNET TELEPHONY Excellence Award
2008-10-09 - TELES VoIP and Mobile Gateways receive Siemens HiPath ready certificate
2008-10-09 - Squire VOIP SS7 Media Gateway provides UK operator Stour Marine with interconnect solution
2008-10-07 - Massive, 900 port Asterisk deployment case study released by Avanzada7 & Redfone (PDF-alert)
2008-10-06 - First iPhone VOIP Application in iTunes App Store
2008-10-04 - Microsoft restores VOIP to Microsoft Live Messenger
2008-10-03 - Vertu Signature S Design luxury dual-mode handset has been released
2008-10-01 - Bandwidth.com Launches 8x8 - SIP Broadband Videophone
ATL IP400 SIP Videophone (Seen working with a GXV3000)
AuPix - SIP & H.323 Videophone with Browser
Grandstream GXV3000 - Cheap H.264 SIP Videophone
InnoTele - Broadband SIP/H323/3G videophones for IP video telephony with WiFi
InnoMedia - MTA 3368 IP VideoPhone
Leadtek Research Inc. - Broadband (SIP/H.323) / ISDN / POTS videophone (site is dead / 2007.06.02)
Ojo - SIP Broadband H.264 Videophone (formerly branded/distributed by Motorola)
Viseon - VisiFone SIP compatible Video Phone
WookSung Elec. - SIP Broadband Videophone, Web Videophone.

(seems like GrandStream GXV-3000 is the only widely available hardware SIP video phone today / 2007.06.02)

Software Phones
Ekiga - Free SIP Audio/Video phone
WengoPhone - Videophone from Wengo
TABLETmedia iFon - soft videophone for PDAs and Smartphones
Kapanga - Free VideoPhone
Video Processing - VOIP video processing technologies and mobile programming.
X-Lite: free soft videophone
Xten eyeBeam - eyeBeam soft
Click here for a FREE trial account!

Click here to go to our website...

Pricing
Wholesale, High Volume Users, PBX Resellers and Termination/Origination Carriers can visit our website to receive a custom price quote by filling out an inquiry form or just click here.

Promotional pricing available for first time users, sign up with $100 or more you will receive an additional $10 credit to your account. First time users only.

Channels
All accounts can support up to four simultaneous calls (any combination of incoming and outgoing). Additional channels may be purchased at $20/month. Customers needing 20+ channels, please contact us for discount information.

Phone Numbers
Phone numbers are $11/month and include free incoming minutes. There is a $25 port fee to transfer your number to VoicePulse. Customers needing 10+ phone numbers, please contact us for discount information.

Incoming Calls
Incoming calls to a U.S. phone number (assigned by VoicePulse, or ported by you) are free.

Incoming Toll-Free Calls
Incoming calls to a U.S. toll-free phone number (assigned by VoicePulse, or ported by you) are 4.9¢/min.

Outgoing US & North American Calls
VoicePulse offers "tiered" rates to most of the United States. This means different areas of the US may have different rates.

Click here for our US/Canada rates

Outgoing International Calls
Click here for our international rates

Downloading Our Rates
Our rates may be downloaded inside the Account Center. Also, customers can use FlexRate to easily lookup our rates in realtime!

Features
Supported Protocols

   * Inter-Asterisk Exchange 2 (IAX2)
   * Session Initiation Protocol (SIP)

Supported Codecs

   * G.711ulaw
   * G. ...

Asterisk config agents.conf

gaetan — Wed, 15 Oct 2008 18:48:49 +0100

agents.conf

ACD distributes incoming calls to the agents of a Queue. Agents are configured in the file queues.conf:

New in Asterisk v1.2.0: The default for ackcall has been changed to "no" instead of "yes" because of a bug which caused the "yes" behavior to generally act like "no". You may need to adjust the value if your agents behave differently than you expect with respect to acknowledgement.

Global options

[agents]
;
; Define autologoff times if appropriate. This is how long
; the phone has to ring with no answer before the agent is
; automatically logged off (in seconds). Please note that if
; this value is greater than the timeout value in your queue
; that agents will not be removed!
;
autologoff=15
;
; Define ackcall to require an acknowledgement by '#' when
; an agent logs in over agentcallpark. Default is "yes".
;
;ackcall=yes
;
; Define wrapuptime. This is the minimum amount of time when
; after disconnecting before the caller can receive a new call
; note this is in milliseconds.
;
;wrapuptime=5000
;
; Define the default musiconhold for agents
; musiconhold => music_class
;
musiconhold => default
;
; An optional custom beep sound file to play to always-connected agents.
;
;custom_beep=beep
;
; Group memberships for agents (may change in mid-file just)
;
;group=3
;group=1,2
group=1
;prevent agent to hangup call by pressing *

thanks to davevg on bug 13707 http: //bugs.digium.com/view.php?id=13707

;encall=no

Groups can be used in queues.conf to specify a member: In this way all incoming calls in the queue will be forwarded to the agents of this agent group. Below we will give you an example how to call an agent group:

member => Agent/@1

or

member => Agent/:1

Both notations have the same effect, the queue will call the agent group with number 1

Recording options

This section is devoted to recording agent's calls
The keywords are global to the chan_agent channel driver

; Enable recording calls addressed to agents. It's turned off by default.
recordagentcalls=yes
;
; The format to be used to record the calls (wav, gsm, wav49)
; By default its "wav".
;recordformat=gsm
;
; Insert into CDR userfield a name of the the created recording
; By default it's turned off.
;createlink=yes
;
; The text to be added to the name of the recording. Allows forming a url link.
;urlprefix=http://host.domain/calls/
;
; The optional directory to save the conversations in. The default is
; /var/spool/asterisk/monitor
;savecallsin=/var/calls

Agents definition

this last section contains the definition of the agents.

Syntax:

agent => agentid,agentpassword,name

agent => 3000,1234,Santiago Ruano Rincon
agent => 4002,1234,Maria Lucia Muñ¯º Grass
agent => 4003,1234,Diego Mauricio Paz

Asterisk Agents
Asterisk cmd AgentLogin
third party way using HylaFax and OPAL to send and receive fax through Asterisk 1.4. See also rejected patch 12931 that includes a T.38 gateway. Attractel offers a commercial T.38 gateway solution for Asterisk.
In Asterisk 1.6 also origination and termination features will be added (with gateway functionality still missing)

FAQ

Q: Does Asterisk support T.38 ATAs behind NAT with canreinvite=no ?
A: See below

Q: Can I terminate T.38 calls to PSTN with Asterisk T.38 passthrough(via Zaptel)?
A: yes, http://bugs.digium.com/view.php?id=12931
CallWeaver supports T.38 termination and gateway operation.

ATA COMPATIBILITY

ATA1(CALLER)	FIRMWARE	NAT	ASTERISK VERSION	ATA2/GW(CALLED)	FIRMWARE	NAT	works?	notes
HT496	1.0.3.44	yes	SVN TRUNK 40000	HT496	1.0.3.44	yes	yes	example
HT496	1.0.3.64	yes	SVN-branch-1.4-r53152	Patton SN4960	T4	no	yes*	sometimes bad quality(ht496 unregister after fax is sent)
SPA2100	3.3.6	no	Branch 1.2+patches	Cisco AS5300	IOS 12.3	no	yes	example
Kapanga	2152b	no	SVN-branch-1.4-r47911	Kapanga	2152b	no	yes*	*receiving kapanga crash
Kapanga	2156	yes	SVN-branch-1.4-r53152	Patton SN4960	T4	no	yes
Patton SN4524	3.20	yes	Asterisk 1.4.1	Patton SN4524	3.20	yes	yes
SPA2102	5.1.6	yes	Asterisk 1.4.2	Patton SN4960	R4.1	no	NO
SPA2102	5.1.1	yes	Asterisk 1.4.2	Patton SN4960	R4.1	no	yes
SPA2102	5.2.5	yes	Asterisk 1.4. ... Asterisk High Availability Solutions spamblock — Wed, 15 Oct 2008 17:21:20 +0100 Ways to increase system availability and balancing: DNS SRV on the CPE side but not all phones handle this. SARK-HA from Aelintra Telecom offers High Availability Asterisk out-of-the box. Runs Aelintra's SARK UCS MVP Asterisk implementation on a pair of servers.... Real-time failover takes less than 20 seconds to complete. Setup requires only 4 additional data fields to filled out in the SARK globals panel. Illustrated set-up guide HERE. Ranch Networks offers High Availability White_Paper_one_one_HA.pdf solution for Asterisk. This is Hardware based solution. (Just for two asterisks boxes). Flip1405 Manages virtual IP between two Asterisk servers and queries UDP5060 for state changes Downtime less than 30 seconds Only 2 dependencies (nmap and arping) Incredibly easy to setup SERVERware. Fault tolerant and high availability solution with unlimited scalability. Commercial Failover switches to automatically switch connections (T1, Ethernet, etc.) to a backup system. CSS: You can make load-balancing with failover with multiple asterisk Altéon : A better tool with permit to load-balance RTP but there is problem is you use qualify=yes and nated phones Big-IP: You can make load-balancing with failover with multiple asterisk (coming soon the real SIP proxy functionalities) Ask me if you have questions about layers 7 switchs Vovida has a SIP load balancer. This allows several Asterisk servers to be setup and appear to be a single server to users. Other load balacing approaches involve the SER SIP proxy, UltraMonkey (see below) or simple DNS round-robin. And then there's also app_distributor as third party application or app_random. there are a lot of bugs and the last version was released in 2002 Page Contents ALABAMA ALASKA ARIZONA ARKANSAS CALIFORNIA COLORADO CONNECTICUT DELAWARE WASHINGTON, DC FLORIDA GEORGIA HAWAII IDAHO ILLINOIS INDIANA IOWA KANSAS KENTUCKY LOUISIANA MARYLAND MASSACHUSETTS MICHIGAN MINNESOTA MISSISSIPPI MISSOURI MONTANA NEBRASKA NEVADA NEW HAMPSHIRE NEW JERSEY NEW MEXICO NEW YORK NORTH CAROLINA NORTH DAKOTA OHIO OKLAHOMA OREGON PENNSYLVANIA PUERTO RICO RHODE ISLAND SOUTH CAROLINA TENNESSEE TEXAS UTAH VIRGINIA WASHINGTON WEST VIRGINIA WISCONSIN WYOMING ALABAMA Asteria Solutions Group http://www.asteriasgi. ... RolePlayingTwoPhonesTwoRooms proppy — Wed, 15 Oct 2008 15:23:22 +0100 HOWTO: Role Playing Game with Phones using Asterisk and other FLOSS Problem Description This HOWTO describes, in the context of a role playing game, a set of four phones used by the players for communication during the game. The game takes place in a hotel or a conference center and the players are in two separate rooms. When the players arrive at their location, the recording phones are plugged in and tested. The game duration is six hours maximum. There is no cable connection between the two rooms. Each conversation is recorded in separate timestamped files and are available in real time, on a file system or through a standard networked protocol. The phones are never used for conferences between more than two phones. The hardware and software solution is designed to work simply and reliably. The installation does not require a technical person and the probability of a malfunction within one game session is less than 1 time out of 10. The key details presented in this HOWTO are: The hardware that should be acquired (phones and computer to store the recordings and network the site), which software should be installed and which software needs to be developed to glue the solution together. A description of how to use the solution created, from installation to putting the hardware back in a box. The user is assumed to be completely clueless as to how the technology works. Hardware Recommendations This section describes which hardware should be used to set up the game. One Box in Each Room? A key requirement here is that "there is no cable connection between the two rooms". This makes wireless routers a key component in the system, and as such, the strong inclination for an Asterisk hacker will be to use the wireless router itself as the Asterisk server. The process for doing so is talked about in many places, including this wiki. However, I do not recommend this solution in this particular case, for the following reasons: The boxes are not particularly powered, and doing full recording of the game might have gaps, and/or cause slowdown of the networks. Configuration and installation of Asterisk is much more difficult on these boxes, and often the latest version of Asterisk is not available. The boxes do not have enough memory without serious hacks (to make SD card attachment possible) to hold the recordings for the full event easily. Server Boxes This leaves basically two options: (a) buy a larger full-fledged computer capable of being both the wireless router and the Asterisk instance, or (b) buy separate wireless router and Asterisk servers. Option (a) is appealing, but if your goal is to keep the bulkiness of the hardware to a minimal, this can simply be courting disaster. The reason is that the small boxes that I plan recommend for this task use laptop-style components, and can be flakey when you mix-and-match two many tasks at once. Therefore, I am recommend the solution of (b). I mentioned a number of small boxes that are on the market. Forschung-Direkt and MIXvoip. Target goals and features: LGPL-licensed core, GPL GUI Decoupled Signaling, Storage and Visualization aspects Modularized, lightweight core layer GCC4 ready code Single codebase for Linux, Windows and MacOS SQLite as default storage backend QT4.4 as GUI frontend Webkit integration Even simpler (than old Kiax) to use UI Completely brandable Remote configuration Simplified integration with service providers (via JSON) Support for multiple service providers Support for simultaneous calls Registry fail-over support Live CDR and Contacts search Codecs: G711, iLBC, GSM, Speex Noise reduction filter I18n support Kiax ver 2 has been released on 24.09.2008. Binary builds are available for Windows, Linux and Mac OS on SourceForge. http://sourceforge.net/projects/kiax. Home Page: http://www.forschung-direkt.eu/projects/kiax2 Kiax 0.8.x (legacy, unsupported) Kiax is an IAX softphone that uses iaxclient for IAX communication and QT3.x (for Linux) and QT4 (Windows) as GUI. Despite of the "K" at the beginning of its name, Kiax runs well NOT ONLY ON KDE, but also on other desktop environments. There is a port of its latest version (0.8.51) to Windows based on the GPL version of QT4 Features: Multiple registration accounts Multiple call appearances Hold/resume call Call Transfer Noise reduction Echo cancellation AGC Supports ULAW, GSM, iLBC, Speex Grouped Contact List DTMF Dialpad Call Register User-friendly Settings Dialog Customizable User Interface Desktop integration Direct IAX URL dialing possible Command-line dialing possible Command execution on incoming call International Support - English, Czech, German, French, Polish, Bulgarian, Brasilian Portuguese, Macedonian, Malay, Italian, Spanish, Hebrew Current version: 2.0 Legacy version: 0.8.51 NOTE: Please update your Kiax to version 0.8.51 - it fixes security flaw in iaxclient's code, as well as fixes amd64 md5 authorization problem. ...

VOIP-info.org Wiki Changes

Hanlong Technology

voip-info.org

Welcome to the VOIP Wiki - a reference guide to all things VOIP

NEWS

Software Phones

Pricing

Channels

Phone Numbers

Incoming Calls

Incoming Toll-Free Calls

Outgoing US & North American Calls

Outgoing International Calls

Downloading Our Rates

Features

Supported Protocols

Supported Codecs

Asterisk config agents.conf

agents.conf

Global options

Recording options

Agents definition

FAQ

ATA COMPATIBILITY

Asterisk High Availability Solutions

Page Contents

ALABAMA

Asteria Solutions Group

RolePlayingTwoPhonesTwoRooms

HOWTO: Role Playing Game with Phones using Asterisk and other FLOSS

Problem Description

Hardware Recommendations

One Box in Each Room?

Server Boxes