WSRCC Phone Directory

Phones

To dial a WSRCC extension from an Internet connected SIP phone dial XXXX@wsrcc.com where XXXX is the extension number. If your phone allows alpha-numeric entry, you may also dial the person's email address. That should make the phone "number" easier to remember.

Note: some phones come with SRV processing turned off. You have to turn it on before your phone will be able to find the sip server for "wsrcc.com".

SIP Test Numbers

There are no longer any public test numbers. Some people started calling all our desk phones (in numerical order!) after getting bored with calling the test numbers. This was starting to be quite annoying especially when you have to run to get the phone and then you hear a "testing testing" and a hasty "click" as the caller quickly hangs up. Calling in to pickup voicemail and wading through 10's of silent voicemail messages wasn't much fun either. Enough.

Users that can do ISN (ITAD Subscriber Number) based dialing via freenum.org can reach us via the *308 suffix. Faculty, students and staff at the universities involved with the Internet2 SIP.edu trials should be able to call these ISN/ITAD numbers. Several large VOIP providers already provide access to their users.

XXXX *308

For example, from FreeWorldDialup one would dial the ISN/ITAD access number **012 then the WSRCC extension number and finally the wsrcc.com ISN/ITAD suffix *308.

**012 XXXX *308

Similarly, from sipbroker one would dial the ISN/ITAD access number *012 then the WSRCC extension number and finally the wsrcc.com ISN/ITAD suffix *308. (Note: this entry shows up in Sipbroker's sip-proxy list as Tello's public ISN/ITAD helper machine, public.tello.com .)

*012 XXXX *308

You may also dial us directly from sipbroker via *364.

*364 XXXX

Freenum, SipBroker and FreeWorldDialup are all free services that can help folks with traditional 12-button VOIP phones dial a telephone number containing a domain name.

Direct SIP URL Peering

(Sometimes called Peer-to-Peer SIP)

Folks running asterisk servers that want to use the same prefix dial codes that sipbroker uses, but would like to run things directly from their own server can grab this extension.conf file snippet exten-peers.conf or the raw ascii mapping file sip-peers.txt. The dial-out.conf file contains the macro needed by the above conf file.

Folks with asterisk servers should definitely add their servers to sipbroker's list so they can join the fun of direct over-the-net dialing.

Soft-Phone

One open source soft-phone known to work well for calling our asterisk SIP server is ekiga (formerly "gnomemeeting"). It is included in Fedora Core-5 (FC5) and is located at Applications->Internet->IP Telephony in the stock user menu.

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wolfgang.rupprecht+web@gmail.com (Wolfgang S. Rupprecht)
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last updated $Date: 2007/05/24 22:21:55 $ ..