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.comwrote: 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.commth...@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/
Re: [sipx-users] HA with Failover problems
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.commailto: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.commailto:mth...@socaltelephone.com?subject=reply%20from%20email%20footer ___ sipx-users mailing list sipx-users@list.sipfoundry.orgmailto: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.netmailto:tgrazi...@voice.myitdepartment.net Fax: 434.984.8431 Email: tgrazi...@myitdepartment.netmailto:tgrazi...@myitdepartment.net LAN/Telephony/Security and Control
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.comwrote: 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
Re: [sipx-users] HA with Failover problems
Tony, I am sorry that I must have sounded like I was inferring that you intentionally frustrated me. :) Of course you weren't trying to. 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 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? We do plan to have somewhere around 2,000 users once we get up and rolling. 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)? 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? I am getting so confused, it is crazy. 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.commailto: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.orgmailto:sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.orgmailto: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
Re: [sipx-users] HA with Failover problems
With 2000 users you will most certainly run into a BLF issue with 4.2.1 because of a known issue with the RLS server. I have a patch RPM for the 64 bit version of sipX 4.2.1 if you require it to hold you over until 4.4 comes out. From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Mark Theis Sent: Friday, September 17, 2010 2:45 PM To: Discussion list for users of sipXecs software Subject: Re: [sipx-users] HA with Failover problems Tony, I am sorry that I must have sounded like I was inferring that you intentionally frustrated me. :) Of course you weren't trying to. 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 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? We do plan to have somewhere around 2,000 users once we get up and rolling. 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)? 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? I am getting so confused, it is crazy. 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.commailto: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.orgmailto:sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.orgmailto:sipx-users-boun...@list.sipfoundry.org
Re: [sipx-users] HA with Failover problems
On Fri, Sep 17, 2010 at 3:44 PM, Mark Theis mth...@socaltelephone.comwrote: 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
Re: [sipx-users] HA with Failover problems
That would be wonderful. I am currently using the 32 bit version in the VM but when I install on the real servers, it will be the 64 bit version. Any idea of when 4.4 will be released? Thanks! From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Josh M. Patten Sent: Friday, September 17, 2010 12:51 PM To: Discussion list for users of sipXecs software Subject: Re: [sipx-users] HA with Failover problems With 2000 users you will most certainly run into a BLF issue with 4.2.1 because of a known issue with the RLS server. I have a patch RPM for the 64 bit version of sipX 4.2.1 if you require it to hold you over until 4.4 comes out. From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Mark Theis Sent: Friday, September 17, 2010 2:45 PM To: Discussion list for users of sipXecs software Subject: Re: [sipx-users] HA with Failover problems Tony, I am sorry that I must have sounded like I was inferring that you intentionally frustrated me. :) Of course you weren't trying to. 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 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? We do plan to have somewhere around 2,000 users once we get up and rolling. 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)? 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? I am getting so confused, it is crazy. 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.commailto: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
Re: [sipx-users] HA with Failover problems
On Fri, Sep 17, 2010 at 3:58 PM, Tony Graziano tgrazi...@myitdepartment.net wrote: On Fri, Sep 17, 2010 at 3:44 PM, Mark Theis mth...@socaltelephone.com 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. If you find it, please let me know, I'm currently struggling with this myself on 4.2.1 install. I have a wireshark where when i shutdown primary, the secondary node gets contacted then returns 404 and I cannot figure out why. Relevant DNS is this === hubler.us. IN NAPTR 2 0 s SIP+D2T _sip._tcp hubler.us. IN NAPTR 2 0 s SIP+D2U _sip._udp finch IN A 192.168.1.173 _sip._tcp IN SRV 1 0 5060finch _sip._udp IN SRV 2 100 5060finch _sip._tcp.rr.finch IN SRV 1 0 5070finch _sip._udp.rr.finch IN SRV 2 100 5070finch parrot IN A 192.168.1.171 _sip._tcp IN SRV 1 0 5060parrot _sip._udp IN SRV 2 100 5060parrot _sip._tcp.rr.parrot IN SRV 1 0 5070parrot _sip._udp.rr.parrot IN SRV 2 100 5070parrot Proxy log on secondary node's proxy == 2010-09-17T21:00:32.733957Z:13125:SIP:ERR:finch.hubler.us:SipRouter-11:41E9A940:SipXProxy:SipUserAgent::send outgoing call 1 2010-09-17T21:00:32.735080Z:13126:SIP:WARNING:finch.hubler.us:SipSrvLookupThread-19:41A84940:SipXProxy:DNS query for name 'rr.finch.hubler.us', type = 1 (A): returned error ha-not-working.pcap Description: Binary data ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] HA with Failover problems
This is the document I was thinking about. Ignoring the views part of it, you simply get to the meat of the matter... http://wiki.sipfoundry.org/display/xecsuser/DNS+Concepts+for+sipXecs The document title is not what the whole thing is about. Here you see the RR (resource record) for sip1/sip2/sip3 weighted differently but the SIP (both udp and tcp) all has the same priority, so there is no preference set on that. In this example the SIP SRV records all have the same priority. The document explains views, but no matter how you do it, you can set a different priority for DNS based on the value (1,2,3, 10, 20, 30, 100,200,300, it is just a weight). The DNS must also be stated to be MASTER on the MASTER and SLAVE on any HA members (/etc/named.conf). So internally you can create views that say if you come from this network A I want you to have sip1 as first priority and if thats not there sip2 is second. If you come from network B I want to change the priority around. This is how you would do qwasi load balancing based on DNS, etc. example.com.IN NS sip1.example.com.example.com. IN NS sip2.example.com.example.com.IN NS sip3.example.com. example.com.IN NAPTR 2 0 s SIP+D2T _sip._tcp.example.com.example.com. IN NAPTR 2 0 s SIP+D2U _sip._udp.example.com. _sip._tcp.example.com. IN SRV 1 0 5060 sip1.example.com. _sip._udp.example.com. IN SRV 1 0 5060 sip1.example.com. _sip._tcp.example.com. IN SRV 1 0 5060 sip2.example.com. _sip._udp.example.com. IN SRV 1 0 5060 sip2.example.com. _sip._tcp.example.com. IN SRV 1 0 5060 sip3.example.com. _sip._udp.example.com. IN SRV 1 0 5060 sip3.example.com. _sip._tcp.rr.sip1.example.com. IN SRV 1 0 5070 sip1.example.com. _sip._tcp.rr.sip1.example.com. IN SRV 2 100 5070 sip2.example.com. _sip._tcp.rr.sip1.example.com. IN SRV 3 100 5070 sip3.example.com. _sip._tcp.rr.sip2.example.com. IN SRV 1 0 5070 sip2.example.com. _sip._tcp.rr.sip2.example.com. IN SRV 2 100 5070 sip1.example.com. _sip._tcp.rr.sip2.example.com. IN SRV 3 100 5070 sip3.example.com. _sip._tcp.rr.sip3.example.com. IN SRV 1 0 5070 sip3.example.com. _sip._tcp.rr.sip3.example.com. IN SRV 2 100 5070 sip1.example.com. _sip._tcp.rr.sip3.example.com. IN SRV 3 100 5070 sip2.example.com. On Fri, Sep 17, 2010 at 5:02 PM, Douglas Hubler dhub...@ezuce.com wrote: On Fri, Sep 17, 2010 at 3:58 PM, Tony Graziano tgrazi...@myitdepartment.net wrote: On Fri, Sep 17, 2010 at 3:44 PM, Mark Theis mth...@socaltelephone.com 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. If you find it, please let me know, I'm currently struggling with this myself on 4.2.1 install. I have a wireshark where when i shutdown primary, the secondary node gets contacted then returns 404 and I cannot figure out why. Relevant DNS is this === hubler.us. IN NAPTR 2 0 s SIP+D2T _sip._tcp hubler.us. IN NAPTR 2 0 s SIP+D2U _sip._udp finch IN A 192.168.1.173 _sip._tcp IN SRV 1 0 5060finch _sip._udp IN SRV 2 100 5060finch _sip._tcp.rr.finch IN SRV 1 0 5070finch _sip._udp.rr.finch IN SRV 2 100 5070finch parrot IN A 192.168.1.171 _sip._tcp IN SRV 1 0 5060parrot _sip._udp IN SRV 2 100 5060parrot _sip._tcp.rr.parrot IN SRV 1 0 5070parrot _sip._udp.rr.parrot IN SRV 2 100 5070parrot Proxy log on secondary node's proxy == 2010-09-17T21:00:32.733957Z:13125:SIP:ERR:finch.hubler.us: SipRouter-11:41E9A940:SipXProxy:SipUserAgent::send outgoing call 1 2010-09-17T21:00:32.735080Z:13126:SIP:WARNING:finch.hubler.us: SipSrvLookupThread-19:41A84940:SipXProxy:DNS query for name 'rr.finch.hubler.us', type = 1 (A): returned error What is the SOA on finch? Is the named.conf setup as SLAVE? ___ 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:
Re: [sipx-users] HA with Failover problems
(that is normal I think). This is a 2 part question. Looks like I will need to add my A records for www in here... AND... the real question... should either of my servers be able to do the web management/customer interface Or should the www A record point only to the Primary server? Thanks again!!! From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tony Graziano Sent: Friday, September 17, 2010 2:25 PM To: Discussion list for users of sipXecs software Subject: Re: [sipx-users] HA with Failover problems This is the document I was thinking about. Ignoring the views part of it, you simply get to the meat of the matter... http://wiki.sipfoundry.org/display/xecsuser/DNS+Concepts+for+sipXecs The document title is not what the whole thing is about. Here you see the RR (resource record) for sip1/sip2/sip3 weighted differently but the SIP (both udp and tcp) all has the same priority, so there is no preference set on that. In this example the SIP SRV records all have the same priority. The document explains views, but no matter how you do it, you can set a different priority for DNS based on the value (1,2,3, 10, 20, 30, 100,200,300, it is just a weight). The DNS must also be stated to be MASTER on the MASTER and SLAVE on any HA members (/etc/named.conf). So internally you can create views that say if you come from this network A I want you to have sip1 as first priority and if thats not there sip2 is second. If you come from network B I want to change the priority around. This is how you would do qwasi load balancing based on DNS, etc. example.comhttp://example.com.IN NS sip1.example.comhttp://sip1.example.com. example.comhttp://example.com.IN NS sip2.example.comhttp://sip2.example.com. example.comhttp://example.com.IN NS sip3.example.comhttp://sip3.example.com. example.comhttp://example.com. IN NAPTR 2 0 s SIP+D2T _sip._tcp.example.comhttp://tcp.example.com. example.comhttp://example.com. IN NAPTR 2 0 s SIP+D2U _sip._udp.example.comhttp://udp.example.com. _sip._tcp.example.comhttp://tcp.example.com. IN SRV 1 0 5060 sip1.example.comhttp://sip1.example.com. _sip._udp.example.comhttp://udp.example.com. IN SRV 1 0 5060 sip1.example.comhttp://sip1.example.com. _sip._tcp.example.comhttp://tcp.example.com. IN SRV 1 0 5060 sip2.example.comhttp://sip2.example.com. _sip._udp.example.comhttp://udp.example.com. IN SRV 1 0 5060 sip2.example.comhttp://sip2.example.com. _sip._tcp.example.comhttp://tcp.example.com. IN SRV 1 0 5060 sip3.example.comhttp://sip3.example.com. _sip._udp.example.comhttp://udp.example.com. IN SRV 1 0 5060 sip3.example.comhttp://sip3.example.com. _sip._tcp.rr.sip1.example.comhttp://tcp.rr.sip1.example.com. IN SRV 1 0 5070 sip1.example.comhttp://sip1.example.com. _sip._tcp.rr.sip1.example.comhttp://tcp.rr.sip1.example.com. IN SRV 2 100 5070 sip2.example.comhttp://sip2.example.com. _sip._tcp.rr.sip1.example.comhttp://tcp.rr.sip1.example.com. IN SRV 3 100 5070 sip3.example.comhttp://sip3.example.com. _sip._tcp.rr.sip2.example.comhttp://tcp.rr.sip2.example.com. IN SRV 1 0 5070 sip2.example.comhttp://sip2.example.com. _sip._tcp.rr.sip2.example.comhttp://tcp.rr.sip2.example.com. IN SRV 2 100 5070 sip1.example.comhttp://sip1.example.com. _sip._tcp.rr.sip2.example.comhttp://tcp.rr.sip2.example.com. IN SRV 3 100 5070 sip3.example.comhttp://sip3.example.com. _sip._tcp.rr.sip3.example.comhttp://tcp.rr.sip3.example.com. IN SRV 1 0 5070 sip3.example.comhttp://sip3.example.com. _sip._tcp.rr.sip3.example.comhttp://tcp.rr.sip3.example.com. IN SRV 2 100 5070 sip1.example.comhttp://sip1.example.com. _sip._tcp.rr.sip3.example.comhttp://tcp.rr.sip3.example.com. IN SRV 3 100 5070 sip2.example.comhttp://sip2.example.com. On Fri, Sep 17, 2010 at 5:02 PM, Douglas Hubler dhub...@ezuce.commailto:dhub...@ezuce.com wrote: On Fri, Sep 17, 2010 at 3:58 PM, Tony Graziano tgrazi...@myitdepartment.netmailto:tgrazi...@myitdepartment.net wrote: On Fri, Sep 17, 2010 at 3:44 PM, Mark Theis mth...@socaltelephone.commailto:mth...@socaltelephone.com 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. If you find it, please let me know, I'm currently struggling with this myself on 4.2.1 install. I have a wireshark where when i shutdown primary, the secondary node gets contacted then returns 404 and I cannot figure out why. Relevant DNS
Re: [sipx-users] HA with Failover problems
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. 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. 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. 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. Thank you! -Mark From: sipx-users-boun...@list.sipfoundry.orgmailto:sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.orgmailto: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.commailto: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
Re: [sipx-users] HA with Failover problems
Would you happen to have any notes about the design that you worked on before? Software/hardware names, model numbers, anything like that? I am interested in your solution to SBC that will account for the clients as well. Currently I am spinning my wheels, not know which way to go. The Junipers can be redundant, we have 1 in Florida and 1 in Cali. If we can find the solution in software that will allow the failover to retain current calls and also allow new registrations and calls... that would be my preferred solution. Thanks! From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tony Graziano Sent: Friday, September 17, 2010 4:15 PM To: Discussion list for users of sipXecs software Subject: Re: [sipx-users] HA with Failover problems On Fri, Sep 17, 2010 at 6:36 PM, Mark Theis mth...@socaltelephone.commailto: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.orgmailto:sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.orgmailto: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
Re: [sipx-users] HA with Failover problems
.mydomain.com.IN A 208.79.55.44 ; A record for sip2.mydomain.com ; sip2.mydomain.com. IN A 208.79.55.41 ns2.mydomain.com.IN A 208.79.55.41 mydomain.com. IN A 208.79.55.41 On an semi-unrelated note… I am planning on using “mydomain.com” and “ www.mydomain.com “ as the web interface (that is normal I think). This is a 2 part question. Looks like I will need to add my A records for www in here… AND… the real question… should either of my servers be able to do the web management/customer interface…. Or should the www A record point only to the Primary server? CNAME for www and an A record for the domain. BUT, you'll need to address other domain needs to determine now whether or not the system should be in a subdomain or not. Thanks again!!! *From:* sipx-users-boun...@list.sipfoundry.org [mailto: sipx-users-boun...@list.sipfoundry.org] *On Behalf Of *Tony Graziano *Sent:* Friday, September 17, 2010 2:25 PM *To:* Discussion list for users of sipXecs software *Subject:* Re: [sipx-users] HA with Failover problems This is the document I was thinking about. Ignoring the views part of it, you simply get to the meat of the matter... http://wiki.sipfoundry.org/display/xecsuser/DNS+Concepts+for+sipXecs The document title is not what the whole thing is about. Here you see the RR (resource record) for sip1/sip2/sip3 weighted differently but the SIP (both udp and tcp) all has the same priority, so there is no preference set on that. In this example the SIP SRV records all have the same priority. The document explains views, but no matter how you do it, you can set a different priority for DNS based on the value (1,2,3, 10, 20, 30, 100,200,300, it is just a weight). The DNS must also be stated to be MASTER on the MASTER and SLAVE on any HA members (/etc/named.conf). So internally you can create views that say if you come from this network A I want you to have sip1 as first priority and if thats not there sip2 is second. If you come from network B I want to change the priority around. This is how you would do qwasi load balancing based on DNS, etc. example.com.IN NS sip1.example.com. example.com.IN NS sip2.example.com. example.com.IN NS sip3.example.com. example.com. IN NAPTR 2 0 s SIP+D2T _sip._tcp.example.com. example.com. IN NAPTR 2 0 s SIP+D2U _sip._udp.example.com. _sip._tcp.example.com. IN SRV 1 0 5060 sip1.example.com. _sip._udp.example.com. IN SRV 1 0 5060 sip1.example.com. _sip._tcp.example.com. IN SRV 1 0 5060 sip2.example.com. _sip._udp.example.com. IN SRV 1 0 5060 sip2.example.com. _sip._tcp.example.com. IN SRV 1 0 5060 sip3.example.com. _sip._udp.example.com. IN SRV 1 0 5060 sip3.example.com. _sip._tcp.rr.sip1.example.com. IN SRV 1 0 5070 sip1.example.com. _sip._tcp.rr.sip1.example.com. IN SRV 2 100 5070 sip2.example.com. _sip._tcp.rr.sip1.example.com. IN SRV 3 100 5070 sip3.example.com. _sip._tcp.rr.sip2.example.com. IN SRV 1 0 5070 sip2.example.com. _sip._tcp.rr.sip2.example.com. IN SRV 2 100 5070 sip1.example.com. _sip._tcp.rr.sip2.example.com. IN SRV 3 100 5070 sip3.example.com. _sip._tcp.rr.sip3.example.com. IN SRV 1 0 5070 sip3.example.com. _sip._tcp.rr.sip3.example.com. IN SRV 2 100 5070 sip1.example.com. _sip._tcp.rr.sip3.example.com. IN SRV 3 100 5070 sip2.example.com. On Fri, Sep 17, 2010 at 5:02 PM, Douglas Hubler dhub...@ezuce.com wrote: On Fri, Sep 17, 2010 at 3:58 PM, Tony Graziano tgrazi...@myitdepartment.net wrote: On Fri, Sep 17, 2010 at 3:44 PM, Mark Theis mth...@socaltelephone.com 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. If you find it, please let me know, I'm currently struggling with this myself on 4.2.1 install. I have a wireshark where when i shutdown primary, the secondary node gets contacted then returns 404 and I cannot figure out why. Relevant DNS is this === hubler.us. IN NAPTR 2 0 s SIP+D2T _sip._tcp hubler.us. IN NAPTR 2 0 s SIP+D2U _sip._udp finch IN A 192.168.1.173 _sip._tcp IN SRV 1 0 5060finch _sip._udp IN SRV 2
