Re: [asterisk-users] Polycom Power Specs
501 - 12V, 1A and a power/data cable 601 - 24V, 0.5A 650 - 24V, 0.5A - Dave On Jan 3, 2007, at 11:48 AM, Peder @ NetworkOblivion wrote: Does anybody happen to know the input power specs for the Polycom IP 500 and IP 600? We've mixed up our power supplies and we've got a whole box of them and can't figure out which go to the Polycoms. I would rather not kill the phones by trying random ones ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Looking for voice over IP products? Visit our VoIP store at http:// voipstore.atacomm.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to connect two asterisk server
Your best bet is to contact Sysmaster support at [EMAIL PROTECTED] or 1877-900-3993. I was talking to one of our contacts there and he said that it would be best to have you contact them. In order to get it to work for you they need to know the exact configuration you are trying to set up. We've worked with Sysmaster for some time now and they are very nice and helpful people. -David Looking for voice over IP products? Visit our VoIP store at http:// voipstore.atacomm.com On Jan 1, 2007, at 8:53 PM, Noah Miller wrote: Hi Again Dan - Its a VOIP Switch based on SIP Proxy. Its Voice Master from SysMaster. VoiceMaster only authenticates IP and cant have username password based authentication which asterisk can do. So i need to take some traffic from VoiceMaster to Asterisk and terminate it. That shouldn't be a problem. You can just create a sip friend/peer without a username or password, and with a host=ipaddress statement. Like this in your sip.conf file: [NoAuth-VoiceMaster] type=friend context=your context host=IP Address Of Voice Master disallow=all allow=codecs you want to allow As an addendum to this, it would be a very good idea to make certain that you've properly secured your asterisk server so you're not going to have unwanted unauthorised access. I would probably only do this if your asterisk server is not accessible from the outside world via sip. - Noah ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] spa300 password recovery
Unfortunately I don't think there is a way to just do a password recovery. You can, however, do a factory reset. That would reset the username and password to factory defaults as well as all other settings. To perform a factory reset you need to access the IVR menu via a telephone connect to the unit. When the phone is off hook enter to get to the IVR menu. When in the menu enter 73738 #. Then unit should then ask you to enter 1 to confirm the reset. This will clear ALL the settings on the unit. ~Dave On Dec 22, 2006, at 1:24 PM, Walter Willis wrote: how to recovery password spa3000? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Looking for voice over IP products? Visit our VoIP store at http:// voipstore.atacomm.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: AW: [asterisk-users] Snom or Cisco Phones?
"Polycom with backite LCD (Is there any?)"The Polycom 650 has a backlit display. They wont be shipping until some time this December.On Nov 2, 2006, at 7:51 PM, Zeeshan Zakaria wrote:Cisco is out of question because as somebody already said in this thread, they come with only half of the stuff, and then they are VERY propriatery. They'll give you really hard time in configuration, firmware upgrading, support etc. I'd say CISCO are not made for open source VoIP industry. My suggestion will be one of the Snom 360, Aastra 480i, Aastra 9133i or Linksys 942, or a Polycom with backite LCD (Is there any?)___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Looking for voice over IP products? Visit our VoIP store at http://voipstore.atacomm.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Looking for Wireless Heaset for Polycom 501
The bottom of the page has wireless headsets that the manufacturers say should work with polycom phones. I know customers tend to lean toward the CS50.http://voipstore.atacomm.com/Shops/Browse.aspx/27934028032-50843149312.htmPlantronics has a "Find my headset" feature:http://www.plantronics.com/north_america/en_US/productfinder/usage.jhtmlGnNetcom has something similar:http://www.gnnetcom.com/US/EN/Misc/CompatibleGNProducts.htm?id=13947name=POLYCOMhope this helped,DaveOn Oct 25, 2006, at 2:38 PM, Andrew Joakimsen wrote:I've used the Plantronics ones, similar to these: http://www.plantronics.com/north_america/en_US/products/cat29880043/cat29880058/prod5510016 and they work very well with the headset lifter, The range is pretty good too.However there are more elegant and complete solutions, with those headsets you need to be by the phone to see who is calling and to use the keypad. On 10/25/06, Jim Freeze [EMAIL PROTECTED] wrote: HiI am looking for a good wirless headset to use with the Polycom Soundpoint 501phone. I would greatly appreciate hearing from anyone with good experienceswith a specific device.Thanks-- Jim Freeze___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Looking for voice over IP products? Visit our VoIP store at http://voipstore.atacomm.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] tdm2400p question
The cable is an Amphenol Cable. This may help some.On Oct 16, 2006, at 8:34 AM, Giorgio Incantalupo wrote:Hi Lito,you need a particular cable to connect TDM2400 (which has 1 big port) to a patch panel. Try Google on internet for a retailer.Giorgio IncantalupoLito Lampitoc wrote: I see, thank you very much for all your answers. Btw, the interface looks different than the ordinary rj45, so how are you going to plug in the rj45 plug to it?On 10/16/06, * George Pajari* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: The TDM2400P supports up to six quad modules -- each quad module supports EITHER four FXS ports OR four FXO ports... THEREFORE with 6 quad FXO modules one has 24 FXO ports, with 5 quad FXO modules and 1 quad FXS module one has 20 FXO ports and 4 FXS ports... the remainder of these examples is left as an exercise for the reader. The board does not have to be fully populated (i.e. you do not need to have all six quad module positions filled). g. -- George Pajari, netVOICE communications 604 484 VOIP (484 8647 x102) Open Source VoIP/Telephony Specialists 1 877 NET VOIP (638 8647 x102) www.netvoice.ca http://www.netvoice.ca www.ip-centrex.ca http://www.ip-centrex.ca www.digium.ca http://www.digium.ca www.grandstream.ca http://www.grandstream.ca www.sipura.ca http://www.sipura.ca www.snom.ca http://www.snom.ca ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Looking for voice over IP products? Visit our VoIP store at http://voipstore.atacomm.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] reloading agents and queues
I think it's just "reload"On Aug 14, 2006, at 3:23 PM, Jordan Novak wrote: Is there a manger command that will reload these two configs, something like extensions reload, so it doesn't drop calls in progress. Jordan Novak Senior Telecommunications Engineer Logistics Health Inc.___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Looking for voice over IP products? Visit our VoIP store at http://voipstore.atacomm.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
