In an effort to serve you better, Telephonyware has moved to a new, larger office in Concord, California. We are now located at 1220 Diamond Way in Suite 130, just off Willow Pass Road in between Interstate 680 and Highway 242. The larger space is a significant step in the growth of our company and allows us to stock more product for your convenience.

On a side note, if you like scuba diving or rock climbing, our neighbors offer a good excuse to visit us or Will Call your order. Telephonyware is located adjacent to Touchstone Indoor Rock Climbing Gym and Nautilus Acquatic SCUBA school.

If you have any questions or would like directions, please do not hesitate to Contact Us.

Aastra Telecom is pleased to announce the availability of a significant new software update for the 5i series of SIP telephones – 53i, 55i, 57i, and 57i CT

Release 2.1 delivers many new features and functionalities, offering increased security and flexibility while enhancing user interfaces and simplifying deployment.
This release includes:

- Full interoperability with Sylantro’s SIP-B feature set

- Transport Layer Security (TLS) and Secure Real Time Transport Protocol (SRTP) support

- Extended support for 5i Expansion Modules:

- Aastra 536M is now supported by the 53i, in addition to the 55i, 57i and 57i CT.

- Aastra 560M is now supported by the 55i, in addition to the 57i and 57i CT

- Additional downloading protocols supported via DHCP option 66: HTTP, HTTPS, FTP and TFTP

- Further interoperability enhancements for Broadfsoft’s Broadworks® platforms

- Enhanced usability allowing quick speed-dial setup via Phone User Interface

- Additional XML enhancements including new options and functionalities for XML objects and softkey management

The new firmware update and documentation are freely available from the Aastra website. Please visit aastratelecom.com, click on “Support” and then select “Download Area”.

Recently, Rhino has announced it’s Echo Cancellation solution to it’s full line of analog cards and a technology partnership with Texas Instruments (TI) and Adaptive Digital Technologies, Inc. (Adaptive Digital). Combined with Rhino’s 4-port, 8-port or 24-port analog boards, the product line becomes the RxFxx-EC, a low-cost but powerful insurance policy against echo for the small PSTN-connected telephony system.

Rhino seems to believe there is no reason not to add echo cancellation to your analog setup and we agree. Starting at $329.00 for the 4-port FXO model (R4FXO-EC) it’s an extremely cost-effective tool to mitigate echo on analog lines. Rhino’s existing digital cards will also soon all accept the Rhino Echo Cancellation (REC) module.

The Rhino EC analog cards feature a control element to remove all “bit banging” on the PCI bus, this Rhino-only feature significantly reduces PCI and CPU overhead, something some of their competitors are still working out.

Every Rhino product comes with a 5 year standard warranty and a life time of support, an unparalleled level of surety in their product and a stand-out feature in the thickening competition among the makers of analog PCI telephony care manufacturers.

To review or purchase the existing Rhino analog line please follow this link to Telephonyware.com.

In a hard won effort by the Senior Sales Manager of Linksys, the SPA94x series phones are now upgradable to 4-lines without the need for paying for an upgrade license key.

Now available for download on Linksys.com is the version 5.1.10 firmware for the SPA942 and the version 5.1.8 firmware for the SPA941. Both firmware upgrades feature the update to use all four lines available on the phone at no cost. To download the new firmware, from the Linksys.com homepage, go to “Support”, “Technical Support” and then “Choose a Product”. Find the “Voice over IP (VoIP)” category and select the model of phone you wish to upgrade from the drop-down menu. As always, read the directions first before attempting installation.

In this blogger’s humble opinion, this should have happened from the start.

Aastra Telecom has announced the availability of a new software update for their line of SIP telephones including Models 480i, 480i CT, 9133i, 9112i, 53i, 55i, 57i, and 57i CT.

Key highlights of the new software include:

• Additional XML flexibility with new options and functionalities for XML objects.
• Enhanced VLAN functionalities that enable the phones to pass untagged packets unmodified to the PC port.
• Improved Audio features that provide additional benefits for visually impaired users.
• Advanced incoming call and Intercom handling options when the user is already on the phone.
• Extended SPRE support for 9112i, 9133i and 53i phones.
• Additional features available with the 5i family, to allow the configuration of Directed Call Pickup with a BLF or BLF List softkey.
• Improved features for failover conditions.
• Further interoperability enhancements for Sylantro®, Asterisk®, BroadfSoft® and MetaSwitch based VoIP environments.

The new firmware update and documentation are freely available from the Aastra’s website. Please visit www.aastratelecom.com

click on “Support” and then select “Download Area”.

Polycom SoundPoint(R) IP and SoundStation(R) IP Software Release SIP 2.1.1 RevC

Version 2.1.1RevC 4 May 2007

The new Polycom software can be downloaded by our customers from our ftp site below. Be sure to check out the complete list of changes in the release notes PDF.

ftp://ftp.telephonyware.com/firmware/polycom/

This release adds support for the SoundPoint IP320 and IP330 products. There is also a single issue (#35913) that was addressed on SoundPoint IP430, 550 and 650 products. All other product software is unchanged from the original release SIP2.1.1.

WARNING: The Server Redundancy Behavior in SIP2.1 has changed from that implemented in prior releases. Prior to SIP 2.1 the reg.x.server.y parameters (see section 4.6.2.1 of the SIP 2.0 Administrator’s Guide) could be used for fail-over
configuration The older behavior is no longer supported. Customers that are using the reg.x.server.y. configuration parameters where y>=2 should take care to ensure that their current implementations are not adversely affected. For example the phone
will only attempt advanced SIP features such as Shared Lines, Missed Calls, Presence with the Primary Server (y=1). Refer to Tech Bulletin TB5844 SIP Server fallback for more details.

For more information, refer to Section 4 in the documents included in the zip file for the new SIP image .

The OpenPCI series is Voicetronix’ new entry level CT hardware platform offering four and eight Loop-Start (FXO) or Station (FXS) ports on a PCI 2.2 bus interface running on Linux. It is suitable for low density PC based telephony applications including:

• IP-PBX and PC-PBX systems.
• Call Recording/Logging.
• Interactive Voice Response (IVR), Auto Attendant and Voicemail systems.

The OpenPCI hardware supports a number of open source telephony applications, providing developers with a powerful and considerable set of tools to quickly build a telephony solution for the price of a PC and the OpenPCI card. These applications include:

• Asterisk(tm), a complete open source PBX software platform providing all of the features you would expect from a PBX, and can interoperate with almost all standards-based telephony equipment. Coupled with the OpenPCI you can build a “Hybrid PBX”, giving your office connectivity to the analog PSTN, analog handsets and the flexibility of either VoIP for handsets or VoIP trunks using SIP, IAX or H.323.
• CT Server, a highly flexible client/server library written in Perl designed to be a rapid application telephony development platform for Voicetronix hardware. It is suitable for PBX, IVR, autoattendant and voicemail applications. CT Server, coupled with a Voicetronix built web GUI admin tool is called OpenPBX, not to be confused with the Asterisk-fork project of the same name.
• Logger, a voice recording/logging server configurable using web based GUIs and catering for analogue line interfaces through the Voicetronix OpenPCI and OpenLog cards.

The OpenPCI connects to the analog PSTN via two RJ-45 ports on the back of the card. The use of a pair of CAT-5 cables, split into their 4 constituent pairs, can be crimped with RJ-11 heads or punched down directly into a punch-down block such as the Dynacom 66 Block as sold on the Telephonyware website. Available as a purchase option are 2 break-out boxes, roughly the size of a deck of cards, cleanly convert 2 CAT-5 cables into 8 RJ-11 female interfaces.

Pricing and availability of the Voicetronix OpenPCI can be found here at Telephonyware.

Polycom Phones Daylight Savings Adjustment

The following information was taken from: Polycom SIP Admin Guide 2.0 Section 2.1.7 – Daylight Saving Time Changes

For daylight saving time, dates will be changing in North America in 2007. To compensate for this adjustment, refer to the Polycom “Technical Bulletin 17803: Daylight Savings Time Changes for 2007 on SoundPoint® IP Phones”

Please see page 3 of the Admin Guide for Changes to default values and the Interpretations for parameters in the sip.cfg configuration file.

For updated Firmware, Bootrom and Admin Guide, you can download from our domain here: ftp://ftp.telephonyware.com/firmware/polycom/

Two of the current Talkswitch models have undergone a mild marketing update. The Talkswitch model known previously as the 48CA is now called the Talkswitch 480vs. The model previously known as the 48CVA is now the 484vs. There are no other functional or cosmetic differences. The name of the Talkswitch 24CA shall remain the same. Below is a summary of the major features of the 480vs and 484vs. A full list of features for each model can be found by clicking on the model headings below.

Talkswitch 480vs

4 anlog trunk lines
8 local extensions (analog or SIP)
10 remote extensions

Talkswitch 484vs

4 analog trunk lines
4 VoIP (SIP) trunks
8 local extensions (analog or SIP)
10 remote extensions

The SIP 2.0.1 release features a significant number of product enhancements, bug fixes and many ‘industry firsts’. Most notably the following major new features are included in this software release:

Transport Layer Security (TLS) to allow service providers and enterprises to ensure that their SIP signaling traffic is secure from malicious eavesdropping or interception.

Busy Lamp Field (BLF) support when used with Broadsoft Broadworks Release 13. This feature allows customers to deploy phones for use by Receptionist and Administrators enabling them to monitor the line status of multiple users from their phone. This feature is best utilized by customers using a SoundPoint IP 601 plus Expansion Module(s).

Integration with Microsoft ® Office Live Communications Server 2005 platform. This feature allows telephony features and Presence features to be implemented on the phone when used with the LCS2005 Server. This may be done as a second registration on the phone with the Microsoft ® Office Live Communications Server 2005 platforms to allow a rich Presence solution in conjunction with Microsoft Office Communicator clients in addition to a primary registration with another SIP call server.

Product Enhancements to allow more flexibility on phone configuration and management designed to facilitate a Low/Zero Touch deployment of Polycom SoundPoint IP phones.

The new Polycom firmware and bootroom can be downloaded by our customers from our ftp site below. Be sure to check out the complete list of changes in the release notes PDF.

ftp://ftp.telephonyware.com/firmware/polycom/