The VoIP Users Conference gathers weekly on Fridays to discuss all things VoIP. Free, and open to the public, participants can listen live or download archived recordings.

There are three main ways to access the live conference:

1. via SIP directly
2. via HiDef SIP directly
3. via PSTN (see below for the number)
4. via the Talkshoe client

VoIP Users Conference via SIP

Recently, several participants experienced difficulties in connecting to the conference using SIP. The issue dealt with DTMF recognition and prevented the participant from entering the conference number and pin.

Thanks to the power of SIP, this problem can be circumvented using a SIP Header. Adding the SIP Header of Subject: <passcode>22622</passcode><pin>YOURPIN</pin> will bypass the DTMF needs and enter you into the conference automatically.

Using Asterisk, a SIP Header can easily be added to your dialplan. For example, if you wanted to dial *10 to reach the VoIP Users Conference, you would modify your extensions.conf to contain something like:

exten => *10,1,NoOp(VoIP Users Conference Fridays at 12pm EST. Replace YOURPIN with your talkshoe pin)
exten => *10,n,SIPAddHeader(Subject: <passcode>22622</passcode><pin>YOURPIN</pin>)
exten => *10,n,Dial(SIP/talkshoe@vuc.onsip.com)

VoIP Users Conference via Hi Def SIP

If you have an HD Voice / Wideband capable phone, you can connect directly to the conference using g722 at the following SIP URI:

• sip:200901@login.zipdx.com

So, in Asterisk’s extensions.conf, it may look something like this:

exten => *11,1,NoOp(VoIP Users Conference Fridays at 12pm EST. g722 connection)
exten => *11,n,Dial(SIP/200901@login.zipdx.com)

REMEMBER! To use the wideband (g722) bridge, you need a wideband capable phone.

PSTN, Talkshoe, and Web

The VoIP Users Conference meets every Friday at 12pm Eastern Standard Time. More information can be found by following these links:

• VoIP Users Conference
• Talkshoe
• PSTN –> Dial (724) 444-7444 and enter 22622# 1#