VOIP Failover

I am building a small virtualization system utilizing vSphere 6.0. The system will only be running about 10 VMs. I plan to use two ESXi hosts with Essentials Plus.
Unfortunately High Availability is a requirement. Due to the system being small, I'd like to avoid a SAN if possible and use local storage. I know I could use vSphere Replication, but if the host that is running vCenter fails, my understanding is that you're unable to restore the replica.
Is there anyway to achieve HA without using a SAN or shared storage?
Thanks!

I currently send email from an internal server, for example, [email protected] my Windows Server 2008 R2, I did create a TXT record undertestdomain.net with the following:Textv=spf1 a ipv4:192.168.1.251 include:testdomain.net ~allI executedsendmail on one of thetestdomain.net servers:Textsendmail [email protected] < text_fileI received the email in [email protected], but when I check the headers, it says:TextReceived-SPF: neutral (google.com: 208.105.144.34 is neither permitted nor denied by best guess record for domain of [email protected]) client-ip=208.105.144.34;Can someone explain to me what I could possibly be doing wrong?

Similar Messages

Guaranteeing VOIP bandwidth during planned Datacenter failover

Simple QoS/ToS/or CoS on your routers and switches. This will give your voip all the bandwidth it needs, when it needs it, and the rest will go to the lower priority traffic. I always set the qos tag for voip to 6 on all of my devices and everything else is left at default (1). 1 being lowest priority and 7 is highest but usually reserved for core network device communication. Anyone feel free to correct me if I'm wrong, it's been a while since the last time I set this up.

As part of some SAN optimisation I have found myself having some voice quality issues due to inter-site traffic across our 150Mbit metro-e link, the problem here is we can bust up to the link speed depending on other traffic on the metro e.How would it be best to prioritise the cross site VOIP traffic over a metro-e connection in the following scenario?3 internally visible vlans+metroevlanthemultisitetraffic route over 1 main site client/VOIP network 1 spanned server network this includes the VOIP switches 1 DR site client/VOIP networkSIP trunks come in across both sites (they have external pipe across differing providers), all trunks are active, calls can come in over either trunk and sip phones are able to pair to any switch at either site.VMware replication and site recovery traffic flows from one VCenter/VRA/SRM combination to the...
This topic first appeared in the Spiceworks Community

VOIP over broadband - with QoS?

At the moment we use Cisco 2600 routers with 256k BT kilostream connections to our stations, to provide IT and Cisco VOIP. The kilostream connections cost about ?5k/station/year.
We want to stream video for the IT, but the kilostream isn't up to it. We also have BT broadband (20:1 contention), to the station, generally between 2-8M, which costs about ?500/station/year. We can do video over that fine.
My questions are:-
1) I'd like to get rid of the expensive kilostream and just use broadband. However, we have about 10-25 VOIP phones on each station, and we keep being told that broadband could not support that, the latency is too high, the reliability too low, that it needs QoS and you can't get that with this infrastructure etc etc. Can I use Cisco VOIP over broadband to 2600 routers, would QoS work and what about the quality/reliability? If it is possible, any hints as to how?
2) As an alternative, I was wondering if I could have both kilostream and broadband connected to the back of my 2600 and have different services going over the two connections (eg voice over kilostream and IT over broadband)? And if so, maybe I could have failover from one to another? Can I do this, and if so, any ideas how?
I don't know if it helps, but we normally terminate our broadband IPSEC VPNs on our WatchGuard firewall, although we do also have a Microsoft ISA firewall that we could use instead.
Any help you are able to offer would be very gratefully received.
Regards
Eric

At the moment we use Cisco 2600 routers with 256k BT kilostream connections to our stations, to provide IT and Cisco VOIP. The kilostream connections cost about ?5k/station/year.
We want to stream video for the IT, but the kilostream isn't up to it. We also have BT broadband (20:1 contention), to the station, generally between 2-8M, which costs about ?500/station/year. We can do video over that fine.
My questions are:-
1) I'd like to get rid of the expensive kilostream and just use broadband. However, we have about 10-25 VOIP phones on each station, and we keep being told that broadband could not support that, the latency is too high, the reliability too low, that it needs QoS and you can't get that with this infrastructure etc etc. Can I use Cisco VOIP over broadband to 2600 routers, would QoS work and what about the quality/reliability? If it is possible, any hints as to how?
2) As an alternative, I was wondering if I could have both kilostream and broadband connected to the back of my 2600 and have different services going over the two connections (eg voice over kilostream and IT over broadband)? And if so, maybe I could have failover from one to another? Can I do this, and if so, any ideas how?
I don't know if it helps, but we normally terminate our broadband IPSEC VPNs on our WatchGuard firewall, although we do also have a Microsoft ISA firewall that we could use instead.
Any help you are able to offer would be very gratefully received.
Regards
Eric

Gateway Failover with CCM 4 1 3b

Hi
I am having issues with gateway redundancy on CCM 4-1-3b. Basically when the Primary route group gateway fails, the backup gateway does not work. You can not make external calls out of this gateway unless you change the priority of the gateways in the route group manually. The only other time it failsover is if ip connectivity fails between the primary gateway and the CCM. Has anyone had any issues like this or is there a bug with this version of CCM? Failover with RG/RL should be a basic feature with CCM?

Hi Derek,
CCM regards H323 gateways as a black hole. CCM does not have the ability see the busy/fail status of the H323 ports. As you said, the only way CCM would switch to the second gateway, in the Route List, would be if IP connectivity was lost.
I would recommend that you additionally configuring outbound VOIP dial-peers (with a higher preference value) to forward the call from your first gateway to your second gateway when the first gateway is up and, for whatever reason, the outward ports are busy or down.
Regards
Chris

Least Cost Routing VoIP between CME sites - can it be done?

Is it possible to configure LCR between two CME sites? I have two different CME's that are in two different states and would like to take advantage of the PSTN connections at both sites. I'm currently using h323 dial-peers for 4-digit dialing and toll bypass, but am not quite sure how to get site A to dial out site B's ISDN line for PSTN calls.
I'm using CallManager Express at both sites.
Any hints?
Thanks.

Mike,
This can be done. Make the configuration transparent to the IP phone users, so they will just dial the numbers as long distance numbers. Configure redundancy via dial-peers. Two dial-peers will do the trick. The first dial-peer will send the call to the remote gateway for it to be routed out as a local call. If the number dialed is dialed as 91+10 digits, strip the 91 (and the area code if necessary) before sending the call to the remote gateway which should already be configured to route the call out as a local call. The second dial-peer is a failover dial-peer, to be used incase the WAN is down or the remote gateway is not available to route out the call. In this case, the call is sent out the local gateway as a normal long distance call.
The configuration could look somewhat like this:
voice translation-rule 4
rule 1 /^91212/ /212/
voice translation-profile LCR
translate called 4
dial-peer voice 1 voip
description ** LCR Via CME_Site_B **
destination-pattern 91212[2-9]......
translation-profile outgoing LCR
preference 1
dial-peer voice 2 pots
description ** LD to CME_Site_B Via Local PSTN**
destination-pattern 91212[2-9]......
preference 2
forward-digits 11
port x/x
Hope this helps.
Michael.

Proxy architectures not resilient to failover without LB?

Hello,
          The proxy architectures shown in the weblogic documentation (basic and
          multi tier proxy) have clients connecting directly to a bank of HTTP
          servers.
          Later they have them connecting to a load balancer which distributes
          load across the HTTP severs. I undersatnd this second case, and
          failover is handled by the load balancer.
          I don't understand their first proposal. Failover is not guaranteed by
          the browser when navigating to a site that has mutlitple machines mapped
          to the same DNS setting (which is what they are getting at I imagine?)
          Can somebody please explain the actual phsyical setup they are proposing
          there?
          Thanks!
          Q


Yep, I did a factory default reset, no difference.
Hopefully an engineer high enough will recognize this issue and have it fixed in an upcoming firmware release. It's fairly disappointing it doesn't currently support resolving SRV records, it completely breaks services which rely on those types of DNS records, such as LDAP and VoIP from some providers which avail of them.
It would be nice if there was at least an option to disable the built-in DNS resolver so that when a client DHCPs, the router hands them the IP addresses of the ISPs real DNS servers.

Failover setup on RV016

Is it possible to have this setup on RV016?
WAN1: VOIP traffic (either by port or IP) + failover for WAN 2
WAN2: all other traffic + failover for WAN1
WAN3: failover for WAN1 & WAN2 with connection on demand
Thanks.

Has anyone else figured this out? I'm getting it on every machine I upgrade and I have about 18 more to go and would like to fix it.
These are brand new iMac's shipped with Mountain Lion. I turn them on and the third screen asks if you want to transfer items from a backup or another Mac. I choose another Mac. Then use Migration Assistant on the users old machine (running Snow Leopard 10.6.8) and then let it transfer all of their files.
At just about the end of the transfer the new Mac pops up this error and waits until an action is taken. I've tried every password it could have been on the previous users machine to no avial. The only option appears to be hitting cancel and movi ng on but I'd like to know what is breaking when this happens becase the machines exhibit a strange spinning wheel hang when logging in afterwards?
Here's a screen shot of the new iMac and the problem:
Thanks!

E1 (ISDN) Failover between 2 E1 ports in same router

I have two Asterisk based VoIP servers and each has an E1 card installed.
One is a primary and the other a secondary (standby).
I am in the UK and have 1 ISDN E1 line from BT giving us our incoming/outgoing calls. At the moment this is hard wired to our primary however if there is a failure with the primary it's a manual shift to the secondary. I would like to automate this.
I have been looking at dedicated E1 failover units but these are either very hard to come by or horrendously expensive
I've therefore just had a thought but would like some clarification from the Cisco forum as to whether this would work or not.
I have a Cisco 2901 with 2 WIC cards giving 4 E1 ports. Is it possible to configure this to act as an E1 failover?
I'm not wanting to convert anything to IP, I want it all done with E1, so for example:
BT Line in goes to WIC port 1
WIC port 2 has primary
WIC port 3 has secondary
Primary and Secondary can both make outgoing via BT line however only port 2 is receiving incoming from BT, however if there is a loss of service on the primary all incoming from BT then routes automatically to WIC port 3
It all sounds plausible to me but as I said I don't want to do any conversion to IP as there is sensitive modem traffic and would like to minimise this as much as possible.
Any advice would be greatly appreciated
Thanks

Hello,
What you describe is pretty simple to configure from a Cisco point of view.
To do this you would configure two dial peers each of which point to the same destination number bt with different preferences and destination ports. A basic sample configuration is shown below:
dial-peer voice 2 pots
destination pattern 1...
preference 1
voice-port 2
dial-peer voice 3 pots
destination pattern 1...
preference 2
voice-port 3
In the sample config each dial peer points to the destination pattern 1...
The dots are wild cards so the number range would be 1000 to 1999.
Dial peer 2 has the better preference and so will be used if available and will send calls out of port 2.
Dial peer 3 has the worse preference and so will only be used if dial peer 2 is fully utilized and another call arrives (this is not possible in your setup) or if the port used by dial peer 2 is down.
In a real config the port numbers would likely be different and some extra commands would be needed but hopefully this would give you the basics of how this would work.
In this setup the Cisco router itself becomes a single point of failure so I am not sure that it improves things much over just having a single Asterisk server.
I am wondering whether it would be possible to wire both Asterisk servers to the BT PRI circuit and have the interface on the second server shut down.
The second server could then somehow monitor the first server and if it went down it could bring it's own ISDN card up. You could do this with Cisco routers using Embedded Event Manager and I am sure you could do something similar with Linux scripting.
Hope this helps and let us know how you finally solve this issue.
James

Contact Center Failover Solution?

Hi,
We have CiscoCallmanager Publisher/Subscriber and UCCX Server in one site (A). The other sites (B,C) they all use the contact center services from site A. If something happened to site A, we want site B and C can still work. Right now, we only have some integrated service router which is capable of doing callmanager fallback in site B and C. But for contact center, I believe the router can't do the failover. So how can I realized the contact center service failover? Buy and install another UCCX server in site B and C?
I'm new to VoIP stuff. Please help. Thanks a lot.
Lou

Hi Jonathan,
This contact center failover discussion is long time ago. Finally I got the time to give it shot. I'm manually configuring the ephone, ephone-dn and ephone hunt in the router. I noticed I must configure "dial-peer hunt" to specify the dial peer hunt sequence. But how can I use this command appropriately? It has multiple mode to configure. In the normal operation, the router just sent the inbound contact center calls to CTI route point. We have a pots dial peer to match all the incoming call:
dial-peer voice 1 pots
description Incoming Calls
incoming called-number .
direct-inward-dial
We have two queues, when the link to UCCX and UCM is down, I preconfigured two dial-peer to match two different incoming calls like:
/* Style Definitions */
table.MsoNormalTable
{mso-style-name:"Table Normal";
mso-tstyle-rowband-size:0;
mso-tstyle-colband-size:0;
mso-style-noshow:yes;
mso-style-priority:99;
mso-style-qformat:yes;
mso-style-parent:"";
mso-padding-alt:0in 5.4pt 0in 5.4pt;
mso-para-margin:0in;
mso-para-margin-bottom:.0001pt;
mso-pagination:widow-orphan;
font-size:11.0pt;
font-family:"Calibri","sans-serif";
mso-ascii-font-family:Calibri;
mso-ascii-theme-font:minor-latin;
mso-fareast-font-family:"Times New Roman";
mso-fareast-theme-font:minor-fareast;
mso-hansi-font-family:Calibri;
mso-hansi-theme-font:minor-latin;
mso-bidi-font-family:"Times New Roman";
mso-bidi-theme-font:minor-bidi;}
dial-peer voice 120 pots
description incoming call to group1
service aa-group1
incoming called-number xxx1111111
direct-inward-dial
port 0/0/0:23
forward digits-all
dial-peer voice 130 pots
description incoming call to group2
service aa-group2
incoming called-number xxx2222222
direct-inward-dial
port 0/0/0:23
forward digits-all
Since these two new dial-peer coexists with the old "wildcard" dial-peer matching any incoming calls, I need specify the preferences for them to make router select new dial-peer with auto attendent during the failover. Can the "dial-peer hunt" command do this? Thanks a lot for your help. Really appreciated.
Lou

VOIP monitor in partial service

Hi Expert
I have installede a new uccx version 8.5.1 su2 withe the cop file.
What i found that the voip monitor subsystem is in partial service.,the system is in HA
From RTMT loggs i can found that "
3853: May 14 23:28:01.414 GMT-1000 %MIVR-SS_VOIPMON_SRV-3-VOIP_OPERATION_ERROR:VOIP subsystem operation error: Module Name=LRMConnection.readMSG: lrmHost: 20.250.185.68 , lrmPort: 3000,A specific description for a trace= error is: ,Exception=java.io.EOFException
3854: May 14 23:28:01.414 GMT-1000 %MIVR-SS_VOIPMON_SRV-3-EXCEPTION:java.io.EOFException
3855: May 14 23:28:01.415 GMT-1000 %MIVR-SS_VOIPMON_SRV-3-EXCEPTION: at java.io.DataInputStream.readInt(DataInputStream.java:375)
3856: May 14 23:28:01.415 GMT-1000 %MIVR-SS_VOIPMON_SRV-3-EXCEPTION: at com.spanlink.lrm.io.CRSLRMInputStream.readINT(CRSLRMInputStream.java:211)
3857: May 14 23:28:01.415 GMT-1000 %MIVR-SS_VOIPMON_SRV-3-EXCEPTION: at com.spanlink.lrm.io.CRSLRMInputStream.readMHDR(CRSLRMInputStream.java:356)
3858: May 14 23:28:01.415 GMT-1000 %MIVR-SS_VOIPMON_SRV-3-EXCEPTION: at com.spanlink.VOIPMonitor.subsystem.LRMConnection.readMSG(LRMConnection.java:156)
3859: May 14 23:28:01.415 GMT-1000 %MIVR-SS_VOIPMON_SRV-3-EXCEPTION: at com.spanlink.VOIPMonitor.subsystem.LrmVoipManager.recovery(LrmVoipManager.java:444)
3860: May 14 23:28:01.415 GMT-1000 %MIVR-SS_VOIPMON_SRV-3-EXCEPTION: at com.spanlink.VOIPMonitor.subsystem.VoipServerHeartbeatThread.run(VoipServerHeartbeatThread.java:50)
Kindly help in resolving the issue
Regards
RC

Hi RC,
Please try upgrading it to UCCX 8.5 SU3 and check if it solves this issue.
CSCts87264 -VoIP monitor device sync has no failover function
http://www.cisco.com/web/software/280840578/90779/RN_UCCX_851_SU3.pdf
Hope it helps.
Anand
Pls rate helpful posts by clicking on the stars below the right answers !!

Is ebgp suitable for VoIP?

Hello,
There's a need for us to interconnect a third party for interconnecting voice traffic.
For us we have serious problems with the VoIP requirments (99.999% availability, 50ms delay,10^-4 packet loss) in combination with eBGP and its redundancy failover timers.
Even with a VPN interconnection as described in an inter-AS connection as descibed in rfc 2547bis option (b) or option (c) are not capable of archieving the VoIP requirements.
For me now the best we can do in my opinion is running eBGP vpnv4 peering according to rfc 2547b in combination with BFD as a L2 fault detection.
Any comments on this is welcome.
thanks

Hi,
ebgp will not play a role in the design unless you have configuration errors that will make it to suffer of prefix floods and route flapping.
What will influence your sla parameters instead is the quality of circuits and perfomance/stability of routers used.

New FIOS customer with dropped VOIP calls and Internet connection

I am a new FIOS customer. Got my 50/25 connection a week ago, switching from a TWC 6/1 connection. Ever since the new connection, I've had numerous issues.
My VOIP (Ooma) connection constantly drops and re-connects during conversations
I've had random Internet connection losses, which picks up again after a few minutes
My home alarm starts chirping every once in a while
I've contacted Verizon several times due to these problems and have received varying answers with no resolution of the problem.
The first time I spoke with support, the tech logged into my router and changed the WiFi channel saying that would fix the problem. It didn’t.
The second time I contacted them, the tech ran a bunch of diagnostics and said everything looked fine so it must be an IP address conflict with my devices, because I had a couple devices using static IP addresses. He said everythinf should be DHCP and the last two digits could not be higher than 99 (192.168.1.99). He said FIOS does not support 3-digit numbers at the end.
So I changed all my devices to DHCP and ran some online VOIP tests. It showed a packet loss of 2-5% and MOS score of 1 (which is bad). I was still getting dropped connections, so I disconnected all devices and connected just one computer to the router and tested again. I was still getting packet loss.
Then I called support a third time, this time the tech said there were no 2-digit IP restrictions and that he was detecting there was no UPS baterry backup for the ONT which was probably causing the problem, so he dispatched a field tech to my house.
Today the field tech came (same guy as before), he took one look at the box and said it was too close to my Electric meter and the RF from the meter was causing interference to the FIOS connection and resulting in dropped connection.
He moved the ONT to another location and said that should fix it.
Well, I'm still seeing packet loss and low MOS score when I run the VOIP test.
I don't know how much of what the techs are saying is true and how much is made up stuff.
Has anyone had similar issues and have thoughts on solutions or likely causes for dropped VOIP calls and connections? Could RF be causing this?
I thought going from a 6/1 Cable connection to a 50/25 FIOS connection would be awesome, but this has turned out to be a nightmare, and I may have to switch back to cable if the problem is not resolved.
I would appreciate any help.
Thanks!

Don't know where the packet loss is happening. I ran the VOIP test on myspeed.visualware.com and it shows a packet loss of 2-5% at different times and a MOS score of 1.
The report says MOS should be around 4 for good VOIP calls.
The Verizon tech who came to the house just blamed the electric meter box for RF interference and move the ONT farther away.
My concern is that I'm getting different answers from different techs at Verizon.
Regarding IP addresses. The Router shows a DHCP range from 192.162.1.2 to 192.168.1.254 as available for devices on the network. So, if I need to assign a static IP to a device should I use a number below 99 or above 151?
Thanks!

In oracle rac, If user query a select query and in processing data is fetched but in the duration of fetching the particular node is evicted then how failover to another node internally?

In oracle rac, If user query a select query and in processing data is fetched but in the duration of fetching the particular node is evicted then how failover to another node internally?

The query is re-issued as a flashback query and the client process can continue to fetch from the cursor. This is described in the Net Services Administrators Guide, the section on Transparent Application Failover.

Reporting Services as a generic service in a failover cluster group?

There is some confusion on whether or not Microsoft will support a Reporting Services deployment on a failover cluster using scale-out, and adding the Reporting Services service as a generic service in a cluster group to achieve active-passive high
availability.
A deployment like this is described by Lukasz Pawlowski (Program Manager on the Reporting Services team) in this blog article
http://blogs.msdn.com/b/lukaszp/archive/2009/10/28/high-availability-frequently-asked-questions-about-failover-clustering-and-reporting-services.aspx. There it is stated that it can be done, and what needs to be considered when doing such a deployment.
This article (http://technet.microsoft.com/en-us/library/bb630402.aspx) on the other hand states: "Failover clustering is supported only for the report server database; you
cannot run the Report Server service as part of a failover cluster."
This is somewhat confusing to me. Can I expect to receive support from Microsoft for a setup like this?
Best Regards,
Peter Wretmo

Hi Peter,
Thanks for your posting.
As Lukasz said in the
blog, failover clustering with SSRS is possible. However, during the failover there is some time during which users will receive errors when accessing SSRS since the network names will resolve to a computer where the SSRS service is in the process of starting.
Besides, there are several considerations and manual steps involved on your part before configuring the failover clustering with SSRS service:
Impact on other applications that share the SQL Server. One common idea is to put SSRS in the same cluster group as SQL Server. If SQL Server is hosting multiple application databases, other than just the SSRS databases, a failure in SSRS may cause
a significant failover impact to the entire environment.
SSRS fails over independently of SQL Server.
If SSRS is running, it is going to do work on behalf of the overall deployment so it will be Active. To make SSRS Passive is to stop the SSRS service on all passive cluster nodes.
So, SSRS is designed to achieve High Availability through the Scale-Out deployment. Though a failover clustered SSRS deployment is achievable, it is not the best option for achieving High Availability with Reporting Services.
Regards,
Mike Yin
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Help Needed for Resetting VoIP SPA 2100 Codes

Yesterday afternoon, I lost power to my home for a few minutes; until then my phone worked perfectly (for the week that I had it initially installed). I have the Earthlink TrueVoice program with a ZyXEL P 600 series DSL modem and direct Ethernet cable (NO router) to my ATA adapter (Linksys VoIP model SPA 2100). My computer is an HP Pavilion 8755C with Windows Millenium Edition. I have a standard (wire) phone connection. After the power was restored to my home, I tried to use my ATA adapter (Linksys VoIP model SPA 2100) to make a phone call, but there was a five-second delay between when I spoke and when someone on the other end of the line heard me. I contacted Earthlink through online chat and was told to do the following: Dial **** 877778 # 1 to reset the adapter. I did this and nothing positive happened (though I did hear a recorded English woman’s voice repeat the numbers I dialed); I did, however, lose the dial tone and was unable to connect to the Internet with my DSL. So, to conduct more online chat, I re-routed the Ethernet cable to bypass the ATA adapter; I was successful at restoring my Internet, but, obviously, I had NO phone service. More than 8 hours of online chats to Earthlink technical service resulted in failure each time. I swapped ends of my cables numerous times – no success. I removed the power cords to the ATA adapter and to the DSL Modem MULTPILE times for for up to 5 minutes – no success. The lights on the back of the ATA lit properly, but since the first time I tried the **** 877778 # 1 code, the Status light blinks and the light indicating “Phone 1” stays dark (yes, there is astandard phoneline connected to that port). I tried multiple codes to reset the ATA (re-coding instructions are separated by semi-colons): **** 877778 # 1 ; **** 877778 # 1 # ; **** 732668 # ; *** *73738 # 1 . I unplugged the power cord and Ethernet cable (to the modem) on the ATA adapter for about 1 or 2 minutes after trying to reset the code before reattaching everything (power cord always re-attached last). I tried each of these suggestions MULTIPLE times without success. One Earthlink tech wanted more specific information – the MAC address (which I was told is a 12 digit hexa decimal number that starts with zero). I told him what it was. He asked me to try a “ping” test; I was told to click on Start -> Run, type in cmd and hit Enter. A new window would appear and I should type in ping 192.168.0.1 and hit Enter. I never got to the “new window” because after I typed in “cmd” and hit Enter, I received the message “Windows cannot find cmd.” I was told to dial the code * * * * 110 # and to hear my IP address; I heard nothing. I noticed that the last time I tried to dial **** 877778 # 1, I was interrupted before I typed the final “1” because the recorded voice told me “invalid value.” Around 1 a.m. last night I gave up in disgust and went to sleep, disconnecting everything. I have now reconnected the DSL modem and made the direct connection to my computer (the ATA adapter is still disconnected from power and all connecting cables). I am not an idiot and I have some electronics experience (I have wired several home theater systems), but I have little computer experience (and none with VoIP) and right now I am totally frustrated. The tech support at Earthlink is virtually worthless. Please help (and please keep it in simple terms so that an ignorant individual like myself might learn – I need a nice step-by-step method (including times to wait between restarts, what cables/cords to disconnect and the order to disconnect/reconnect them (if that matters) to reset this device) Thanks from a neophyte who is not afraid to be honest and show his ignorance in this area of knowledge.

hi,
i have a SPA-2100 it's a Sipura, and it now asks for a password. I was putting all my settings in for voip provider and now can not get in my admin? i did try the 73738 and that did not work. I was able to put my providers sip and stun all in. is it i'm locked out from my voip provider? If so why would that be, because this is not their device, it's mine? but how do I get back in my admin or can the voip provider I put in, help?
thanks
Message Edited by jokers32463 on 12-22-2008 06:21 PM

VOIP Failover

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