In several functions of IP Office and network equipment management, email is sent from an automated process to a recipient via SMTP. SMTP is a fairly simple protocol that can be emulated via command lines. Using these command lines, setup information can be verified about the server to which the message is being sent. The only tool necessary for this test is telnet. The commands do have to be fairly exact, but are not hard to implement. (more…)
In each Windows server resides a time server compatible with IP Office. There three methods of updating the IP Office time: manually, via Vociemail, or via a timeserver. Using a Windows server is the most universal, since it does not rely on Voicemail Pro being installed. On occasion the update does not happen. The update should run every hour to synchronize the IP Office with the time on the server. In order to debug this process logging can be enabled on the server.
2 easy ways to synchronize the time in an ipo unit to a pc.
1. The IP Office unit will use the vm pro server’s time if you have all 0’s in the time server address.
2. If there is Windows 2003 server that the ipo unit can ping(ie route to) put this in the timer server field - make sure the time is in 24 hour format(control panel->Regional and Language options->Time)
troubleshooting: use system monitor - the time on the unit will appear within the first couple of lines right after the date
Documentation:
http://marketingtools.avaya.com/knowledgebase/ipoffice/mergedProjects/manager/index.htm?manager_howthesystemreceivestime.htm
IP Office 4.2 introduced a new option for Music On Hold. Prior releases had a single choice for External, via the MOH jack, or Internal, via TFTP the holdmusic.wav file. Now up to 4 seperate sources can be defined. The options are documented in the knowledge Base, but the information is a bit scattered. Here we will consolidate some of the docs and overview the options available in 4.2+ firmware. (more…)
VoIP Assessments are an essential step to a successful VoIP implementation. It identifies potential network issues early reducing the costs of project overruns and post-production troubleshooting.
Catalyst Services has created three different offers to fit your project and budget. Each option offers a different level of engineering involvement. The Catalyst technical support team encourages you to include this first step for all of your VoIP projects. Click here below for details.
Synchronize Calls was introduced in 4.0+ with the addition of the Announcements Tab for Hunt Group. The theory behind this option is to conserve voicemail resources used for announcements while a caller is queued in a hunt group. Enabling this feature allows all callers in the queue to hear the same message, at the same time, thus only using one voicemail resource. Read more for limitations and documentation..
On UMS default installs, users may get in the habit of entering just the server name, and not adding the entire sub-directory for accessing the web application. This results in an unexpected page with an Under Construction message or a picture displaying IIS7. Either way it’s not UMS, and could be frustrating for a end user. Changing the root directory to be the voicemail directory will break the application as the paths will break the UMS webservice. A simple solution is to setup a redirect. Read more to see how…
This post will explain how to convert a QE 46xx phone from p2p to H323. In order to obtain the conversion tool you will need a SSO log-in. If you have an SSO log-in go to partner.avaya.com
The new value series 1600 IP sets will be out soon. These sets will be supported in an upcoming 4.2+ release. There was some original debate as to these being released as 1700 series. The knowledge base now confirms that these sets will be 1600 series. Docs are starting to trickle out with details about the sets. These are IP Phones with paper labels much like the old 44xx sets. They have digital displays and reg/green status LEDs. The phones have a much updated look similar to the high end 9600 sets. The display is back lit but monochrome. Read more for links to the documentation.
Voice Compression Modules (VCM) are an important key to allowing the IP Office to handle such a wide range of phone types. These modules act as a bridge between the Digital/Analog phone world and the IP world. Each VCM is able to handle translations between H.323 or SIP signaling and the control plane, as well as be a real time CODEC for translation to the RTP (Real Time Protocol) used for the Audio streams of a VoIP call. They are very similar to a trunk between two systems.