I was perplexed that this did not work for Steve, as I was sure I had done this before. So I needed to confirm the setup, and just finally got around to it. It does work.
I just set it up on a Polycom 550, running 3.2.6. I configured on the phone, under "Call Handling" to 1. Next, on the individual line, I set "Line Keys" to 2, and set "Calls per line key" to 1. From a VVX500 I dialed the extension, first call appears on button one, placed first call on hold with VVX500, placed second call which appeared on button 2. Additionally, while second call was ringing, I did a directed call pickup from a 3cx soft phone. It picked up the call and I was connected to the VVX500 successfully. I find that having a multi-line phone, and having the same line appearance multiple times just makes it easier for the customer when handling multiple calls. It works. I've had it running on every extension at one site with over 40 extensions for over 2 years now. Never had a reported issue with them. From: [email protected] [mailto:[email protected]] On Behalf Of Tony Graziano Sent: Monday, March 05, 2012 7:24 AM To: Discussion list for users of sipXecs software Subject: Re: [sipx-users] Polycom IP560 Multi lines YOU_SHOULD_NOT_DO_THAT. I'm not sure anyone has ever done that, and I would venture a guess to say "it's not generally supported". Why? The same line is acking the same call(s) 4 times from the same device/address, for starters. Should someone every try to do a call pickup from another phone for that line, it would be might impossible. There are other reasons, but that's why the phone can handle more than one call per line (scrolling display, to sleect and choose calls). Remove the other three instances of the same line and ensure you have not changed any values for the call limits per line. It will take multiple calls on the same line out of the box, so to speak. I would suggest not doing that as what you are tring to do is very unusual and you will find yourself somewhat "alone" in troubleshooting any issues (as you have now). On Mon, Mar 5, 2012 at 9:50 AM, Steve <[email protected]> wrote: I'm having an issue with a multi-line (4) IP560. I have all 4 lines set to the same extension on the phone. When I'm on one line and a call comes in to my extension is goes right to VM. I would expect the second (through fourth) line on the phone to ring. I'm on 3.2.6 firmware and 4.3.1 bootrom In the phone registration page I have: lineKeys=4 callsPerLineKey=1 If I change callsPerLineKey = 2, then I get the second call on the same line, not on the second extension listed on the phones softkey extension list. I'm looking to have the phone's second line ring if the person is on the first line, the third to ring if they're on the first and second etc.. What am I missing? =============== Thank You, Steve http://www.AppraisalHost.com 781-214-6250 Appraisal Host Logo On 02/29/2012 12:25 PM, Todd Hodgen wrote: Not a Broadvox Guru for sure, but I've had some good luck with them. Actually, their new fusion platform works fine now also. You have to request the Mitel Option, and ask that they configure your contact in the To field, and everything works great. I've worked with Steve a bit, but I'm not a firewall guru. Always seems to be audio issues once the call is connected. From: [email protected] [mailto:[email protected]] On Behalf Of Tony Graziano Sent: Wednesday, February 29, 2012 4:51 AM To: Discussion list for users of sipXecs software Subject: Re: [sipx-users] Sipx Architecture question Todd is the broadvox guru. Specifically eat you need to be on their legacy platform. You could always provide a packet capture from the firewall. On Feb 29, 2012 7:35 AM, "Steve" <[email protected]> wrote: Thanks.. I have no issue spending money or time figuring this out however, I've yet to find anyone that's gotten this to work with a pfsense firewall and broadvox. I'm open to using another open source fw as well. So, if there is anyone on the list that has gotten this to work (sipx, pfsense fw and broadvox) I'd like to hear from you. Specifically, my issue is inbound calling. Outbound works fine. Steve On 02/29/2012 07:22 AM, Michael Picher wrote: Server can only have one NIC so this is a fail... What you're trying to do here really isn't that hard... If you can't make it work, I'd suggest spending more time to learn about this stuff and make sure you have the proper hardware. Or seek help from somebody who can just set it up for you and make it work. Both of which may involve some $... But, there are only 2 ways to do things. Mike On Wed, Feb 29, 2012 at 7:01 AM, Steve <[email protected]> wrote: This probably isn't specifically a sipx architecture question.. I've been having some issues with NAT, I've been able to get it to work with one firewall (that's underpowered) but not pfsense. I've even gone so far as to start from scratch going to v1x of pfsense and following Tony's postings and importing pieces of Tony's sample config with no success. I've even tried IPCOP with same results. So, I've decided to abandon the whole NAT thing and just move the server to the DMZ. So my question is this, as I have 2 NIC's in my sipx server can I just add the second IP to the server (for the DMZ public address) and then also have the server plugged into the existing phone (non routable) network? Essentially, my phones (all IP) would be completely segregated with no access to the Internet or other networks. The sipx server would have a connection on that network as they would obviously need to communicate with the sipx server. Are there any issues with this config? Do the phones need any other connectivity other than the sipx server, NTP for example? Thanks, Steve _______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- Michael Picher, Director of Technical Services eZuce, Inc. 300 Brickstone Square Suite 201 Andover, MA. 01810 O.978-296-1005 X2015 M.207-956-0262 @mpicher <http://twitter.com/mpicher> www.ezuce.com ---------------------------------------------------------------------------- -------------------------------- Hope to see you at the sipX CoLab! http://www.sipfoundry.org/sipx-colab A gathering for - open source users, eZuce customers & eZuce partners Get the inside track on 4.6 and a glimpse at the future of sipXecs! _______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users/ _______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users/ LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: [email protected] Helpdesk Customers: http://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net _______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users/ _______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- ~~~~~~~~~~~~~~~~~~ Tony Graziano, Manager Telephone: 434.984.8430 sip: [email protected] Fax: 434.465.6833 ~~~~~~~~~~~~~~~~~~ Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 Ask about our Internet Fax services! ~~~~~~~~~~~~~~~~~~ LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: [email protected] Helpdesk Customers: http://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net
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