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The Graveyard - Products No Longer Supported => Routers / COVR => DIR-655 => Topic started by: blips on May 04, 2009, 11:07:53 AM
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I constantly have problems with my VOIP line with the DIR-655. It keeps losing it's connection. If I reboot the router my VOIP line works. I can make calls do anything I want. After a set amount of time if I go to make a call I do not get a dial tone. If I reboot the Sipura adapter for my VOIP line it doesn't do anything. If I reboot the DIR-655 it works again. Any thoughts on how to fix this? Does anyone else have this problem?
I have a A4 on 3.1 firmware.
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who is your ISP & what modem are you using?
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My ISP is AT&T DSL. I have the 3/756 DSL line coming into my house. My DSL modem is a Speedstream 5100. I have the 655 doing the PPOE authentication and not the Speedstream.
The VOIP line worked perfectly with my Linksys RV082 router for a couple of years. I just replace the Linksys with the D-Link DIR-655 about a week and half ago and I have been having this same problem since. If I reboot the 655 it will work with out issue. It is after a few hours of not using it I don't get a dial tone.
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go back to the dir-655 internet setup and change the dial type to always on if it's set to something else. You may also want to set a static QoS entry for it and reserve it's IP address.
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Make sure SIP ALG is turned off.
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Thanks for the tips but I have done all that.
1) I specifically have an IP address so the VOIP adapter. I forward the UDP ports 5160 5161 to that IP address that VoicePulse recommends.
2) I have my connection set as always on.
3) I turned off SIP ALG yesterday and it did not help.
Any other ideas?
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Go into your Firewall Settings in the router and disable SPI and set NAT Endpoint filtering to Endpoint Independent for both UDP and TCP and save settings.
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Keep an eye on active connections in Internet Sessions. Find your ATA keep an eye on the "Time Out" If it reaches 0 then for whatever reason the ata isn't sending a keep alive. UDP sessions time out after 300 seconds. The forwards should prevent this however.
BTW if it helps I have a Pap2 and don't need to forward anything. Think the only change I made was turning off SIP ALG.
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Go into your Firewall Settings in the router and disable SPI and set NAT Endpoint filtering to Endpoint Independent for both UDP and TCP and save settings.
I tried this. Set it at about 3pm. When I got home from work at 5 things were still good. Then I went to my son's baseball game and got home at 8:30pm. No dial tone. Could my DIR-655 be defective?
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Keep an eye on active connections in Internet Sessions. Find your ATA keep an eye on the "Time Out" If it reaches 0 then for whatever reason the ata isn't sending a keep alive. UDP sessions time out after 300 seconds. The forwards should prevent this however.
BTW if it helps I have a Pap2 and don't need to forward anything. Think the only change I made was turning off SIP ALG.
Ok thanks for getting me to look at the Internet Sessions. I saw that my VOIP device had the wrong IP address. When I entered the MAC address to give the VOIP a static IP I had one digit off. Hopefully now that I fixed that it will work.
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Well I have tried all the suggestions above and I'm still having the same problem. After awhile I still do not have any dial tone.
dcurrey, I do have port forwards and my ata seems to still be timing out. What else can I do to prevent this?
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I also set my VOIP device as a priority 1 on the QoS settings. I'm wondering if this will help.
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Double check you are forwarding the correct ports. Should be a to verify what the ata is using by Internet sessions.
I don't recommend playing with Qos rules until you get basic functions up and running. Don't want to complicate things.
Also do you have access to the ATA status page. It may help give a clue as to what is going on. Like if it can't connect or lost its ip.
Not familiar with the sipra but if you do have access look for something like nat keep alive. Also check to see if it uses a STUN server. May or may not.
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Can you confirm.. Did you say you had SIP enabled or disabled?
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Yesterday, I did not have any time to work on this problem as I was not home until late.
dcurrey, I do not have any access to my VOIP adapter's status page or logs. I will have to check with my VOIP provider if they can let me in the adapter to view these things.
Ru-Fi-Oh, I have SIP disabled.
I think I found out the issue but I think I need a better understanding how the Internet Sessions status page works.
It seems to go out when I'm running a bit torrent program. I noticed in the Internet Sessions page that the bit torrent program has pages and pages of connections. As far as I know the 655 can handle 200 simultaneous connections. The Internet Sessions page, if I'm understanding it correctly, has well over 200 connection. I'm guessing the connection to my VOIP adapter is getting dropped even though it is still listed as connected.
For the bit torrent program I forward 2 different ports to 2 different computers. The internet connections using these ports seem to stay open even if I shut down the bit torrent programs. Is that because I'm forwarding them? Is that why they will not time out?
The tests I need to do are:
1) See if after about 8 hours or long if my VOIP adapter still has a connection without running any bit torrent programs.
2) See if I can run my bit torrent programs without forwarding any ports so connection can time out and don't push me to the maximum internet connections that the 655 can handle.
Does anyone successfully run bit torrent programs with a VOIP adapter?
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Correct. If you forward ports they stay open regardless if anything needs them.
Don't normally run torrents. Grab Kubuntu every six months but thats about it. I limit the download and upload speed in the torrent program itself. I am in no rush. I will have to check to see how many connections it allows by default don't think I changed it.
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Ok, so last night I ran my torrent program on two different computers all night. I went to reset the 655 this morning, because I figured the VOIP line would be dead, and what do you know? The VOIP line was still up. Hopefully, it will just keep working but I have the feeling it may be temperamental.
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Problems And Challenges
* VoIP requires use of an Analog Telephone Adaptor, IP telephone or a computer. All of these require power and will not work if there is a power shut down. Solutions can be a UPS back up. POTS works on the power from the central office, works even though there is not power, and may continue to be the choice of people who fear loss of communication during power failures.
* Present online security measures cannot adequately handle VoIP processing requirements and changes in protocols and mechanism will take time for a hassle free, secure voip service. This includes internet vulnerabilities like Denial of service attacks, Phishing, snooping and spoofing.
* Latency (delay) is another problem in connections that are made. You will find that it takes quite some time before you hear the other person’s voice. And if security measures are implemented to provide encryption then the latency factor will increase causing loss in quality of voice. VoIP is a real time service and computing power can speed up a few things this will push up the cost of equipment.
* Misuse of the technology by hackers, advertisers (spam/spit) has been a problem with the internet and will be so with Voice over the internet as well.
* Backward compatibility is another major factor. VoIP protocols do not seem to effectively work with older firewalls and NAT-Network Address Translation that is a part of some LAN and WAN networks. Voip problems; vulnerability of the network is basically a compromise of the security of the network. The problem is also of other equipment and its operation with the present network mechanism and regulations.
* Wi-Fi hotspots offered by PSTN and even other Wireless services offered for Internet is not as secure as it proclaims to be. Many have raised additional security concerns and these have not yet addressed.
* With many tests done on the equipment and connections, another problem is that with ISP’s and that is a relatively small upstream bandwidth. The ISP gives you a greater bandwidth to download files and streaming video. The upstream capabilities are only one tenth of the down stream capacity and may negatively affect voice communication on the net.
* As the number of users increases and every service converges to the internet network. There is bound to be problems with bandwidth for the large number of users. Quality of service and reliability for real time application is a major problem and may cause loss of words in a conversation when there is congestion in the network.
* Interoperability between the various networks poses another big problem with various overlapping standard. Not all companies especially by major players may accept proper interconnection and standards framed.
* Investment into previous technologies and advances in those technologies prevent the adoption of newer technologies. Being accustomed to the old and not wanting to make a change due to various factors is another problem.
* Regulatory development and its impact on IP telephony is still a challenge. IP telephony is unregulated in most countries and regulatory authorities are monitoring the situation closely. Some countries have addressed specific problems like the FCC ruling for direct 911 calls. This is specific to countries where 911 is the number for emergencies and is an issue for the 911 operators. Further details are given below.
by: Call Center Software (http://www.inin.com)