Re: [asterisk-users] Digium IP Phones - Teleworker Capability?
On 06/14/2012 05:23 PM, asterisk users wrote: Is there a detailed application note in the Digium wiki (or anywhere else for that matter) about these implementing features under Asterisk/Switchvox? Not yet, I don't believe. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium IP Phones - Teleworker Capability?
On Thu, 2012-06-14 at 16:23 -0600, asterisk users wrote: > > This is pretty good news, overall. To comment on Kevin's points: > > - The end-to-end encryption is important to us, because > client-ID-sensitive information is part of our environment. Something > like built-in OpenVPN would work for us, if that were an option. > Yealink and I think Aastra phones have OpenVPN built in. We use Yealink with layer 2 tunnels such that the phones have the same configuration, network wise, wherever they happen to be plugged in. No NAT issues ever. > - Being fault-tolerant (of less than perfect DSL and rural-wireless > connections - if the boss is at his cabin, for instance) and being > very user-friendly about it is really important to end users. Minet > has a heart-beat mechanism so that if the connection goes down between > the phone and the switch, the display shows it. Of course, calls get > diverted to voicemail during that period. > Pretty much all SIP phones work that way. > If something is not working in the network, the user is informed about > it, and when it is fixed, everything continues, including button DSS > status updates, voicemail WMI, etc. > Again all phones work that way. > On typical SIP phones, everything looks normal until you go to use it, > then there is no dialtone, or you just get dead-air on the handset). > Which SIP phone have you been using? The ones we are familiar with - Polycom, Linksys, Yealink, Snom, Aastra, Grandstream - all show you when the network link is down, and all services return as soon as it comes back up. Even Linksys ATAs at least show you an LED of when the device is registered, though you will just get dead air if you pick up the handset. > Our users are pretty demanding, and want a utility-grade solution that > will always work - for them. > > - > Most of it, I think. Give them a try! > > Is there a detailed application note in the Digium wiki (or anywhere > else for that matter) about these implementing features under > Asterisk/Switchvox? > You could probably find 50 people to help you set such a system up on this list (or more appropriately on -biz). Cheers, j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium IP Phones - Teleworker Capability?
On Thu, Jun 14, 2012 at 4:05 PM, Kevin P. Fleming wrote: > On 06/14/2012 04:57 PM, asterisk users wrote: >> >> We couldn't see anything about this on the Digium site, but maybe >> someone here can comment? >> >> Do the new Digium phones provide good "teleworker" functionality? > > > Yes, I believe they do :-) > > >> The benchmark we're comparing against is the capabilities of Mitel >> 3300 IP systems with Mitel 5330 IP phones (running their proprietary >> MINET protocol), specifically: >> >> a. A Mitel phone can be easily configured for teleworker mode (select >> TW mode and the IP of the gateway server). The phone reboots and it >> is ready to be used (once the Mitel border gateway is set to recognize >> the unit's ID, based on its MAC address, printed on the label on the >> back of the phone). If the phone gets reallocated back to a directly >> connected office environment, a simple reset procedure brings it back. > > > Digium phones can do something similar, and in an upcoming firmware release, > there will even be features available to make this happen on a fairly > automatic basis. > > >> b. You can plug in the phone virtually anywhere. It has a built-in >> tunnelling mechanism providing end-to-end encryption and is very >> tolerant of the network configuration, routers, NAT, etc. > > > Digium phones speak SIP and RTP to the server, just like pretty much any > other SIP phone. They employ many modern NAT traversal techniques and should > work in most network situations. They don't currently provide encryption for > signaling and media, though. > > >> c. If the link between the phone and the gateway goes down, the phone >> will restore itself gracefully and automatically once the network >> function resumes. Absolutely hassle-free to the user. > > > I don't understand this; SIP phones don't require this at all. The phone is > an intelligent device on its own. If there is no network connectivity to the > server, then calls cannot be placed or received, but once connectivity is > restored, operation would be back to normal. > > >> d. Users can be configured to have hot-desk functionality. The phone >> has a default extension assigned, but the user can be set up so that >> they can "log in" to their normal office extension number from >> wherever they are. Their office phone is automatically logged-out and >> goes to its default extension when you log in to a teleworker phone >> (you don't have to log out from it first). Your phone buttons, >> display settings, voicemail WMI and access, (everything) move to this >> new phone, and you can work from your home office, on the road, etc., >> and inbound and outbound calls work just like you were there in the >> office (callerid, etc). > > > Yes, this is supported. > > >> These four features would be a big selling point for us to consider >> moving our organization from Mitel to Digium/Asterisk/Switchvox. >> >> How much of this can be done with Asterisk/Switchvox and, say, the >> Digium D70 phone with dynamic button display? > > > Most of it, I think. Give them a try! > > -- > Kevin P. Fleming > Digium, Inc. | Director of Software Technologies > Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming > 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA > Check us out at www.digium.com & www.asterisk.org > > -- This is pretty good news, overall. To comment on Kevin's points: - The end-to-end encryption is important to us, because client-ID-sensitive information is part of our environment. Something like built-in OpenVPN would work for us, if that were an option. - Being fault-tolerant (of less than perfect DSL and rural-wireless connections - if the boss is at his cabin, for instance) and being very user-friendly about it is really important to end users. Minet has a heart-beat mechanism so that if the connection goes down between the phone and the switch, the display shows it. Of course, calls get diverted to voicemail during that period. If something is not working in the network, the user is informed about it, and when it is fixed, everything continues, including button DSS status updates, voicemail WMI, etc. On typical SIP phones, everything looks normal until you go to use it, then there is no dialtone, or you just get dead-air on the handset). Our users are pretty demanding, and want a utility-grade solution that will always work - for them. - > Most of it, I think. Give them a try! Is there a detailed application note in the Digium wiki (or anywhere else for that matter) about these implementing features under Asterisk/Switchvox? Thanks! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-use
Re: [asterisk-users] Digium IP Phones - Teleworker Capability?
On 06/14/2012 04:57 PM, asterisk users wrote: We couldn't see anything about this on the Digium site, but maybe someone here can comment? Do the new Digium phones provide good "teleworker" functionality? Yes, I believe they do :-) The benchmark we're comparing against is the capabilities of Mitel 3300 IP systems with Mitel 5330 IP phones (running their proprietary MINET protocol), specifically: a. A Mitel phone can be easily configured for teleworker mode (select TW mode and the IP of the gateway server). The phone reboots and it is ready to be used (once the Mitel border gateway is set to recognize the unit's ID, based on its MAC address, printed on the label on the back of the phone). If the phone gets reallocated back to a directly connected office environment, a simple reset procedure brings it back. Digium phones can do something similar, and in an upcoming firmware release, there will even be features available to make this happen on a fairly automatic basis. b. You can plug in the phone virtually anywhere. It has a built-in tunnelling mechanism providing end-to-end encryption and is very tolerant of the network configuration, routers, NAT, etc. Digium phones speak SIP and RTP to the server, just like pretty much any other SIP phone. They employ many modern NAT traversal techniques and should work in most network situations. They don't currently provide encryption for signaling and media, though. c. If the link between the phone and the gateway goes down, the phone will restore itself gracefully and automatically once the network function resumes. Absolutely hassle-free to the user. I don't understand this; SIP phones don't require this at all. The phone is an intelligent device on its own. If there is no network connectivity to the server, then calls cannot be placed or received, but once connectivity is restored, operation would be back to normal. d. Users can be configured to have hot-desk functionality. The phone has a default extension assigned, but the user can be set up so that they can "log in" to their normal office extension number from wherever they are. Their office phone is automatically logged-out and goes to its default extension when you log in to a teleworker phone (you don't have to log out from it first). Your phone buttons, display settings, voicemail WMI and access, (everything) move to this new phone, and you can work from your home office, on the road, etc., and inbound and outbound calls work just like you were there in the office (callerid, etc). Yes, this is supported. These four features would be a big selling point for us to consider moving our organization from Mitel to Digium/Asterisk/Switchvox. How much of this can be done with Asterisk/Switchvox and, say, the Digium D70 phone with dynamic button display? Most of it, I think. Give them a try! -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Digium IP Phones - Teleworker Capability?
We couldn't see anything about this on the Digium site, but maybe someone here can comment? Do the new Digium phones provide good "teleworker" functionality? The benchmark we're comparing against is the capabilities of Mitel 3300 IP systems with Mitel 5330 IP phones (running their proprietary MINET protocol), specifically: a. A Mitel phone can be easily configured for teleworker mode (select TW mode and the IP of the gateway server). The phone reboots and it is ready to be used (once the Mitel border gateway is set to recognize the unit's ID, based on its MAC address, printed on the label on the back of the phone). If the phone gets reallocated back to a directly connected office environment, a simple reset procedure brings it back. b. You can plug in the phone virtually anywhere. It has a built-in tunnelling mechanism providing end-to-end encryption and is very tolerant of the network configuration, routers, NAT, etc. c. If the link between the phone and the gateway goes down, the phone will restore itself gracefully and automatically once the network function resumes. Absolutely hassle-free to the user. d. Users can be configured to have hot-desk functionality. The phone has a default extension assigned, but the user can be set up so that they can "log in" to their normal office extension number from wherever they are. Their office phone is automatically logged-out and goes to its default extension when you log in to a teleworker phone (you don't have to log out from it first). Your phone buttons, display settings, voicemail WMI and access, (everything) move to this new phone, and you can work from your home office, on the road, etc., and inbound and outbound calls work just like you were there in the office (callerid, etc). These four features would be a big selling point for us to consider moving our organization from Mitel to Digium/Asterisk/Switchvox. How much of this can be done with Asterisk/Switchvox and, say, the Digium D70 phone with dynamic button display? Thanks for all comments! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
