On Fri, Sep 17, 2010 at 3:44 PM, Mark Theis <mth...@socaltelephone.com>wrote:

>  Tony,
>
>
>
> I am sorry that I must have sounded like I was inferring that you
> intentionally frustrated me. J  Of course you weren’t trying to.
>

No biggie. Sometimes a picture helps. And I am famous for lines as a
drawing...

>
>
> 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.

>
>
> I know about the DNS SRV records and I think that I have it setup
> correctly, but… maybe I don’t and it is causing me the problems. Would it be
> bad to post my domain name here so someone (who feels like it), could look
> at the DNS records?
>
>
>
Sure. Feel free to change the SIPDOMAIN and IP's (find and replace). There
should be a good example on the wiki. Let me find it and send it to you.


> 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.

>
>
>  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).

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 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.

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.
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