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The CDW part number is: 672205 And the cisco part number is: CON-SNT-CP7960 Hope this helps... In fact I think I will add it to the Wiki. ~c ------------------------------ Message: 11 Date: Mon, 27 Sep 2004 10:40:38 -0500 From: "W. Kevin Hunt" <Kevin@hbcorporate.com> Subject: RE: Cisco Downloads --> was --> Re: [Asterisk-Users] Cisco 7960 andAsterisk...not working... To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Message-ID: <FE3AABFA37FDE645A491291B1FDF1E2B2BF503@hunt02.hbc...

I am the one that placed the bounty. After it being there for 2 months and getting no takers (and very few if any people asking about it), we are almost finished writing it in house. I'll keep the bounty up untill we do finish our product so if anyone beats us to getting it working they'll get paid... W. Kevin Hunt CCIE #11841 MCSE, Linux+ SME www.huntbrothers.com >

An interesting article for those needing ammunition to sell Asterisk within their organisation or to others: "Is open source IP telephony ready for prime time? Yes" by Zenas Hutcheson, St. Paul Venture Capital Network World, 06/07/04 http://www.nwfusion.com/columnists/2004/0607faceoffyes.html On a related note, they also have an article arguing the contrary position (see link within

Hi- I have an upcoming application that requires use of PRI channels that are primarily used for high-volume incoming traffic, but that are to be used for outbound calling as well. Of course, one option is to have dedicated outbound channels reserved, but this is an inefficient use of channel resources. Normally PBX's are designed to have the CPE yield to an incoming call if a particular

After speaking with several people, and even participating in a forum of several other people with echo issues, I thought I'd share what we've done (well actually what our chief R&D engineer, Brett Bourn has done...) First let me say that normal cheapy PC hardware couldn't be made to function with out echo. We tried on both the single port Digium T1 card and the 4 port Digium T1

I have a 3com sip phone that I can't seem to get correctly setup for * I set the identity of the phone to 5kevin2 I set the password on the phone to 222 (not the admin pw to change the phone's settings, but the pw it is supposed to send to the sip server) I set the sip server to my asterisk servers ip. If I don't add anything to the sip.conf file for the phone, I can dial internal

Does anyone have a config they'd like to share w/ the above hardware doing termination for asterisk? I've got one coming in tomorrow along w/ some DSP's and would like to not have to create the config from scratch to start testing. W. Kevin Hunt

You are correct... No glare on a PRI W. Kevin Hunt CCIE #11841 www.huntbrothers.com -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Steven Critchfield Sent: Tuesday, May 25, 2004 3:26 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] "Glare" condition - How well does asteriskhandle?

We have a strange echo problem. Maybe echo isn't the correct term. When we make a call f/ a SIP phone (we have several 7960's, some 3coms, and I've even tried a softphone, all on the same 100BaseTX network) to the pstn, if the person I'm calling has a PRI or channelized T1 f/ Bell, then the sound is perfect, couldn't be better. If I make a call to a person with a plain POTS

can you please share the cdw part # for the $ 10 service contract ? W. Kevin Hunt CCIE #11841 MCSE, Linux+ SME www.huntbrothers.com > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > Christopher Jacob > Sent: Saturday, September 25, 2004 9:51 PM > To: