[Asterisk-Users] Major bug solved in IPSwitchBoard

I have been working on solving a major issue with IPSwitchBoard. It was reported that IPS would use all available memory and get the PC to grind to a halt.

I could not understand this as I had it running on many different PC’s in Denmark.

I now found the bug:

IPS would crash on any PC that had "." configured as decimal point (in Denmark we use ",") this meant that IPS would consume all memory as it tried to make fonts 100 times larger than meant to be. I have now installed a PC with US windows on it just for testing. I hope that I have not caused you too much trouble.

Download the software for FREE: http://ipsoftware.thorben.dk

IPSwitchBoard is totally customizable and will give you, among other things:

• Unattended/attended transfers.
• Park calls and retrieve/forward them again.
• Organize all your SIP, IAX extensions (automatically retrieved from Asterisk).
• Monitor all extensions, queues and Parked Calls.
• Dynamically log extensions in and out of queues.
• Set Do Not Disturb on Extensions and give a reason
• Set Call forwarding for extensions
• Set Dual call for extensions Monitor multiple calls on an Extension/Queue Monitor
• Extension online status
• Totally customizable
• Make you own skins with logo’s etc.
• Make your own buttons – DND, Online, Queue Status, Call Forward, Dual Call, MWI etc.

IPDesigner is a unique tool for customizing and setting up IPSwitchBoard. With IPDesigner you can design your own IPSwitchBoard with Company logo and all the buttons you need for the Operator.

IPSwitchBoard works with .IPS files. These files contain everything needed for IPSwitchBoard such as bitmaps, server and extension configuration. You can build .IPS files with IPDesigner.

Download the manual to read more: http://www.ipdanmark.dk/IPSwitchBoard/IPswitchBoard%20Manual.pdf

Thorben

[Asterisk-biz] AstBil New Billing, Routing and Management software for Asterisk

AstBill is not only a web-based, user-friendly billing interface to Asterisk. It is also a Asterisk configuration and management tool and a standardized implementation of Asterisk using REALTIME and static configuration as you please.

http://astbill.com

AstBill DEMO: http://demo.astbill.com

Here are some of the features of AstBill:

User-friendly End User Web interface gives access to a range of functionality:

- Personal Contact Directory with Categories
- View SIP, IAX and Virtual Accounts
- Virtual Accounts (Lets you forward your calls to any extension you want also time based forwarding)
- Credit Control on outgoing calls
- Show Balance, Expenditure, Payments and number of Calls on each account
- Set warning balance for email when low credit on account
- View Numbers Dialed and add them to the Contact Directory
- View Numbers Dialed by Names from the Contact Directory
- Dynamic International Rate Table(Each customer can have his own price list using Brands)
- Rate Table in his Currency of choice
- Call Data Records including cost of each call and time based billing
- Call Data Records in his Currency of choice
- Switchboard (Displays live status of users phones and ongoing calls)
- Allows one click calling from GUI and direct to phone
- Call Parking sends calls to parking and then redirects to phone
- Allows transfers of calls
- Edit your Account setup
- Asterisk Billing and Management
- Edit voicemail setup including email and pin
- Create Time Based Dialing. Allows you to forward your calls based on time and day.
- Each user can have unlimited of SIP, IAX and Virtual Accounts
- Each user can have unlimited Prepaid Card Accounts linked to his userid
- Specify your hardware and change the viewable name of your accounts
- Temporary disable SIP, IAX or Virtual Account
- Manage your Incoming Public Numbers including Time based forwarding

User-friendly Administrator Web interface gives access to a range of functionality:

- Show Balance, Expenditure, Payments and number of Calls on each account
- Call Data Records including cost and Sales on each call
- Branding Module. Allows you to create Brands in any Currency
- For each Brand define Currency, Billing Increment, mark-up and connection charges
- Flexible Dynamic International Rate Table for Each Brand in any Currency
- Server Status (Displays live status of users phones and ongoing calls)
- Show Peers. List of the last clients(SIP and IAX2) that have connected to the Asterisk server.
- Audit Trail. Show IP, Port and UserAgent for each call
- Manage your Incoming Public Numbers including Time based forwarding
- Manage Trunks. You can use unlimited ZAP, IAX and SIP trunks.
- Time Based Trunk Dialing. Each trunk can have his own time based dialplan
- Temporary disable trunks
- Trunks can be rated after cost. Allow for cost based Dialing
- Define maximum concurrent outgoing calls on each trunk
- If lowest cost trunk is fully used (busy) the system will choose the next available trunk.
- Define unlimited outgoing routes and link them to your trunks
- Store cost of your outgoing route for each trunk for efficient cost control
- All outgoing routes are stored independent on the client price list
- Define customer price lists for each Brand and Currency
- Advanced customer management andportal management
- Integrated E-commerce module and web shop is available under GPL
- Define list of VOIP hardware commonly used
- Full Hardware Inventory. Store mac address and serial numbers of client hardware
- View and Store Customers payments
- Asterisk Billing and Management
- Manage Pre Paid and Post Paid customers. Full Credit control by User Account
- View important Server logs from web interface
- Define maximum concurrent calls on each Customer Account

AstBill includes the functionality needed by most small- to medium-sized businesses (SMB). It is also very efficient as a platform for small VOIP providers. AstBill is used by several small businesses and is the platform for http://astartelecom.com the new European Asterisk call termination service currently in beta.

Are Casilla

[-biz] – FYI: Sipura “brand” no more, now Linksys

As of today, Linksys has made some changes in the wake of their acquisition of Sipura. The current Sipura product line has been rebranded Linksys, there will no longer be ATA's and SIP phones bearing the Sipura name/logo.

Linksys has also retired the SP enrollment form, whereby ITSP's had to fill out and submit paperwork, and be approved by Linksys in order to purchase products like the RT31P2-NA, WRT54GP2-NA and PAP2-NA. This requirement has now gone away.

Saskatchewan = great plains, but no VOIP.

According to Michael Hennessy, President of the Canadian Cable Telecommunications Association, this is because SaskTel, the exclusive telephony provider in the province, is so hostile to the eight-year-old competition-encouraging policies of the Canadian Radio-television and Telecommunications Commission that it refuses to provide VoIP.

The key reason seems to be because they don't feel they have to.

"Eight years after the local competition decision, there is still no local competitor, no Vonage and no number portability in Saskatchewan," Hennessy says in a statement provided for Canadian cable television trade publication Cablecaster magazine.

"SaskTel claims that the competitive safeguards contained in the VoIP and prior decisions are not appropriately applied to Saskatchewan on the grounds that: (a) larger competitors from outside the province would not be price regulated in Saskatchewan; and (b) the CRTC decision will have the effect of keeping consumer prices artificially high in that province," Hennessy writes. "In our view, so long as SaskTel retains 100% of the local voice market, the first claim is irrelevant and the second is simply wrong.

"The only thing standing between consumers and lower prices in Saskatchewan is SaskTel itself," he continues. "Under the CRTC decision, SaskTel can introduce a lower priced VoIP service and expect expedited CRTC approval. Bell Canada received expedited approval for its digital voice service within two weeks of its application.

"With no competition to date," Hennessy concludes, "SaskTel remains firmly in the driver's seat. SaskTel can immediately offer a discounted VoIP service to the general public - before a single local voice competitor enters the province."

stolen from: http://blogs.zdnet.com/ip-telephony/?p=668

New VOIP Speakerphone

seen here: http://mvox.com/mv900.html

New Linux Router – With VOIP support

cool stuff seen here: http://www.linuxdevices.com/news/NS3289921374.html

Broadband customer premises equipment chip vendor PMC-Sierra is shipping a Linux-powered hardware/software reference design for VoIP WiFi broadband routers. The company says high-performance Linux routing helps the design deliver PSTN-like voice quality (public switched telephone network), while simultaneously handling data and video traffic.

According to PMC-Sierra, the reference design includes "all hardware and Linux-based software to build a production-ready product" that can provide wire-speed 100Mbps Ethernet routing and firewalling, and PSTN-like voice quality regardless of other network traffic loads. Additionally, the device is claimed capable of delivering video-over-IP services.

The design is based on PMC-Sierra's MSP4200 "VoIP router on a chip," expected to begin shipping in Q3, 2005. The MSP4200 is a 32-bit MIPS-based SoC (system-on-chip) with integrated voice processor DSP (digital signal processor).

Onboard I/O includes two FXS (foreign exchange subscriber) ports and one FXO (foreign exchange officer) port for integration with analog PBX systems and handsets. Other ports include a WAN Ethernet port, an integrated 802.11a/b/g Wi-Fi access point, and four fast Ethernet ports.

On the software side, the design includes a SIP stack supporting G.711, G.723.1, and G.729 compression, kernel-based routing and firewalling, and management interfaces based on open source application software.

PMC-Sierra says that VoIP technology based on the MSP4200 processor achieved a "mean opinion score" greater than 4.1 in China Netcom Group Labs voice quality tests, as well as certification from China Telecommunication Technology Labs and Korea's Telecommunications Technology Association. PMC-Sierra's VP of marketing, Dino Bekis, said, "Carrier validation of VoIP technology is essential. The China Netcom Group Labs testing has provided our best results to-date."

ZyXEL's assistant VP of CPE, Albert Ju, said, "The residential broadband VoIP market is entering a period of very rapid growth. Carriers are demanding PSTN-equivalent voice service quality, combined with simultaneous high-bandwidth routing performance."

Market research firm ABI said in June that it expects mobile phone makers to increasingly add VoWiFi capabilities to mobile phones, because the technology allows mobile phones to work better indoors.

Availability

PMC-Sierra's VoIP Wi-Fi Router "turnkey system design" is available now, including an MSP4200 processor, schematics, gerber files, sources, object code, and documentation.

PMC-Sierra acquired its MSP (multi-service processor) chip line from Brecis last year, with an eye on the emerging triple-play market. The line was originally launched by Brecis in April of 2003.

Anyone got 1.5 billion? Vonage might be forsale soon.

if you believe what you read here anyway :)..

NEW YORK - Vonage, the private internet telephony start-up, is being encouraged to consider a sale while it presses ahead with plans for an initial public offering.

UBS and Deutsche Bank, the investment banks chosen by Vonage to underwrite its stock market listing, have been suggesting that the company pursue a "parallel process", according to people familiar with the matter.

This would involve seeking out, or responding to, expressions of interest in the company from potential buyers while at the same time moving forward with plans to raise up to $600 million in an IPO.

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