On Fri, Sep 17, 2010 at 6:36 PM, Mark Theis <mth...@socaltelephone.com>wrote:
> In fact… your email is really helping me feel better and less > frustrated. I have spent the last week trying to get this working and I was > fearing that it was all a waste. Now I have hope again! > > > > I am interested in the SBC software for sure. I am not opposed to > offloading work from sipXecs to another hardware/software that can do a > better job. Failover is a must in my situation, whatever I can do to > achieve this I am willing to do. > > > > I'm happy to discuss. Just let me know. There are advantages of using > external SBC's and gateways in sipXecs. Not everyone needs those > advantages. > > > > Perfect! Maybe you can tell me more about your SBC software. Sounds like > it can handle the users as well? I defiantly need to figure this thing out > so that it is redundant with failover. > > We do plan to have somewhere around 2,000 users once we get up and > rolling. > > > > That's an easily attainable number between two systems. > > > > I figured as much. And these servers are beefy. 3U Supermicro, Dual quad > core xeon 2.5ghz with 16gb ram. I would think that they can handle much > more than 2,000 users. > Nice. > > > As far as the SBC goes… To clarify… Can sipXecs do it alone, keeping > the calls active if one of the servers fails (or we drop the Internet for > some reason)? > > > > No. Right now if you use sipXrelay (media relay for remote users) or > sipXbridge (siptrunking), these anchor the media and is not redundant. The > registrar/proxy functions are redundant. (i.e., a failure occurs and the > registration is dropped and re-established to the failover. Once it is > registered it can make calls, etc. A smart enough SBC which can also handle > the remote users does the rest, so redundancy is very possible (with the > right parts and pieces). > > > > Ok… Sounds like you have this all figured out. Do you have this kind of > thing running now? I think that you have a good idea of what I am looking > for… I would think that this is what everybody normally would be looking > for actually. > > I designed one with some help (it was also to use MPLS for branch connections) for a PHONE COMPANY who wanted to sell it to their customer, then they (their customer) got cold feet about open source. So yeah, it's well figured out. > > > Sounds like you are saying that it can’t do it (which is ok and I am fine > with using another process for SBC). We do have 2 Juniper SRX240’s (one is > CA and one in FL) that does SBC… Would this replace the SBC from sipXecs? > > > > It would replace the SBC, I have never tested one (sounds like fun), but it > does not address remote users. > > > > I wouldn’t be using the Juniper switch until I can prove that this will all > work and I install the server on the big boxes. AND… sounds like it leaves > me with a hole in my plan. I need to figure out how to deal with the remote > users. > Well, and can it be redundant? Be fun to see. > > > I am getting so confused, it is crazy. > > > > No biggie. It also becomes important to use a "better" remote user NAT > traversal method if you are supporting a large number of remote users. It > really would blow to have to walk people at a lot of remote sites in making > firewall changes (turning off SPI and SIP ALG) to get a phone registered. > There are better methods to do that with large numbers, but it does require > a budget. > > > > > > Any suggestions for the “better” remote user NAT traversal methods? I > certainly do not want to employ 20 technicians to talk people though setup > 10 hours a day. What kind of budget are we talking? I have a limited budget > but I can request more if I can argue my case well enough. > Yes, an SBC with all the smarts built in will do just that. > > > > > > > Thank you! > > > > -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 12:24 PM > > > *To:* Discussion list for users of sipXecs software > *Subject:* Re: [sipx-users] HA with Failover problems > > > > Nothing was said to intentionally frustrate you. I wanted to make an > example of how the proxy/sipx is not involved in the RTP stream once the > call is established, and also how it IS involved after the call is > established. > > On Fri, Sep 17, 2010 at 3:02 PM, Mark Theis <mth...@socaltelephone.com> > wrote: > > 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. > > > > As I said, the call path and devices between makes a big difference. If the > call is via a trunk and using sipxbridge and that server goes down its going > to drop the call. I'm sorry it frustrates you, but the way in which you > design the system can ensure that this does, or does not, happen. The > components do matter. If all of the users are remote then sipx is anchoring > and/ore relaying their media, so the way to achive this is with an > independent SBC which has a redundancy feature (which exist and can be > used). > > > > 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. > > > > The failover would requires DNS with SRV records which specify priority. > You have not provided a lot of information on whether or not the PRIORITY > has been setup according to the guidlelines (see the wiki or provide some > more information). > > > > 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. > > > > Yes, for registration. Any calls or users on the unavailable server will be > disconnected and re-registered, at which time calling can continue. > > > > > > 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. > > > > I have access to SBC software that will do that, and keep the RTP intact, > but its not open source, and works with sipx too. > > > > In a high volume environment, sometimes its better to remove some roles > from sipx, which can make the system much more flexible. > > > > Thanks for your help! > > > > -Mark > > > > If it were me, and its not, I would approach this with a single server in > each site with a synced SBC for remote users and trunking, with sipx only > having basic roles would certainly do the trick, as long as you are not over > a couple of thousand users and have the hardware spec'd properly... > > > > > > *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> > wrote: > > I am new to this list and also to sipXecs, so please excuse any ignorance > that you might notice. J > > > > 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<mth...@socaltelephone.com?subject=reply%20from%20email%20footer> > > > > > _______________________________________________ > sipx-users mailing list > 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 > Fax: 434.984.8431 > > Email: tgrazi...@myitdepartment.net > > LAN/Telephony/Security and Control Systems Helpdesk: > Telephone: 434.984.8426 > sip: 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. > > > _______________________________________________ > sipx-users mailing list > 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 > Fax: 434.984.8431 > > Email: tgrazi...@myitdepartment.net > > LAN/Telephony/Security and Control Systems Helpdesk: > Telephone: 434.984.8426 > sip: 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. > > > _______________________________________________ > sipx-users mailing list > 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 > Fax: 434.984.8431 > > Email: tgrazi...@myitdepartment.net > > LAN/Telephony/Security and Control Systems Helpdesk: > Telephone: 434.984.8426 > sip: 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. > > _______________________________________________ > sipx-users mailing list > 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 Fax: 434.984.8431 Email: tgrazi...@myitdepartment.net LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: 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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