I appreciate your response. Your answer frustrates me a bit. I keep reading that the servers can go down without losing the active call, but I can't get this to work. I also keep reading that failover does work, but I also can't get this to work. This has left me with more questions, of course.
Since all my users will be remote, I am hosting the sipxecs servers at datacenters. I was hoping to be able to make this work with 1 server in our our California datacenter and 1 in our Florida datacenter. Does this sound like it would work? I am using a sip trunk on the sipXbridge currently. Is it possible to make the trunking of the sip lines redundant? So that on the loss of the Master, the Redundant server assumes that role? If not, I don't see how the pbx could function, except for internal calls to the registered phones. Thanks for your help! -Mark From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tony Graziano Sent: Friday, September 17, 2010 10:24 AM To: Discussion list for users of sipXecs software Subject: Re: [sipx-users] HA with Failover problems Not currently. There is a lot of work ongoing to enhance the redundancy of the system, but the RTP stream itself is not redundant. Right now your "proxy" and "registrar" are redundant. This means if one goes down, phones will register to the available server and new calls will route through the available server. If you are on a call and the proxy breaks it's POSSIBLE your call will stay up, depending upon the media path. Look at this [sipx pstn gateway] <<-->> [SIPX] <<-->>(ETHERNET NETWORK) <<-->> <USER/PHONE> | | -------------------------------------MEDIA PATH--------------------------------------------------------- In the above example, once the call comes in and hits the proxy, the proxy notifies the user. Once the user picks up the phone and the media is established the call goes PEER to PEER. If you are using sip trunks with sipXbridge, the media is anchored in sipx. As a demo, I routinely place a call via an external SBC (not sipXbridge) or a PSTN gateway (or to another internal user (not remote user using sipxrelay on sipxecs), and unplug the ethernet cable for sipx and you will see the call stays up. What might be missed is the nhangup and an accurate CDR record for the call. Hope you liked the book. Sounds like you got a lot done! On Fri, Sep 17, 2010 at 1:14 PM, Mark Theis <mth...@socaltelephone.com<mailto:mth...@socaltelephone.com>> wrote: I am new to this list and also to sipXecs, so please excuse any ignorance that you might notice. :) I am trying to setup a HA pbx. I thought that it was working perfectly until I took the master offline and it doesn't appear that failover works. Since there is not a ton of information available about this (I purchased the Building Enterprise Ready Telephony Systems with sipXecs 4.0 and have done over 8 hours of google search looking for the answer to this problem), I am not sure if I am configuring the system correctly or if my configuration is even to be expected to be failover ready. I have 2 servers configured. 1 is the master and 1 is the distributed server. It appears that DNS is working fine. The distributed server has the following services running: CDR HA Tunnel Redundant SIP Router Shared Appearance Agent Redundant SIP Router Media Relay Redundant SIP Router SIP Registrar Redundant SIP Router SIP Proxy Redundant SIP Router And it also has the "Redundant SIP Router" Server Role. Am I confused about how this is supposed to work? My understanding is that a call in progress should not drop if the master server goes offline and that the Redundant server should take over. I know that I wouldn't have voicemail support at this point, but I am hoping to be able to maintain a call and be able to make additional calls if the Master goes down. Is this even possible? From what I keep reading, it is... but I can find only brief mention of the configuration process. Any help would be appreciated! Mark D. Theis Southern California Telephone and Energy office (951) 693-1880 Ext. 212 fax (951) 693-1550 Cell (951) 545-1013 or (949) 682-VOIP 27515 Enterprise Circle West Temecula, CA. 92590 mth...@socaltelephone.com<mailto:mth...@socaltelephone.com?subject=reply%20from%20email%20footer> _______________________________________________ sipx-users mailing list sipx-users@list.sipfoundry.org<mailto:sipx-users@list.sipfoundry.org> List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- ====================== Tony Graziano, Manager Telephone: 434.984.8430 sip: tgrazi...@voice.myitdepartment.net<mailto:tgrazi...@voice.myitdepartment.net> Fax: 434.984.8431 Email: tgrazi...@myitdepartment.net<mailto:tgrazi...@myitdepartment.net> LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpd...@voice.myitdepartment.net<mailto:helpd...@voice.myitdepartment.net> Fax: 434.984.8427 Helpdesk Contract Customers: http://www.myitdepartment.net/gethelp/ Why do mathematicians always confuse Halloween and Christmas? Because 31 Oct = 25 Dec.
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