I will briefly outline the situation currently and I would like your opinion on the best move for us.
Currently we are running a Trixbox server with VoicePulse and Teliux VOIP services. We have two providers, because we were getting fuzzy calls or dropped called constantly. Having two providers, regardless of what one we use, doesn’t help. We have done many things, such as changing servers, changing ISP’s, having dedicated DSL internet connection only for voice traffic, and nothing has helped. Staticy “are you on a cell phone” type calls both outgoing and incoming.
Something is telling me that the VOIP technology just isn’t there yet or we are overlooking something extremely simple and stupid. I have yet to find this out and cannot continue to have an unprofessional sound on the phone.
We are using Linksys SPA 941 phones and a Cisco 7940 SIP. We LOVE the features Trixbox (Asterisk) offers and the versatility but the reality is that our voice quality is terrible and has to change. So we are in the midst of going to be changing to analog phone lines.
My question really is if we go this route, what would you recommend? There are only three of us here in the office, so I was thinking of bagging the whole Asterisk idea and going with a simple pure analog NEC or Avaya or Talkswitch phone system. Our entire phone system, including phones, would basically have been a waste and would need to be sold.
Another alternative is getting a two or four port FXO card for the Asterisk box to convert the analog signal to digital format. Problem with that, is that I have heard from nearly every Asterisk expert I have talked to that the technology isn’t there and you’ll still get really bad echo occasionally from the analog to digital FXO conversion.
What is your opinion(s) on this matter? I don’t want to purchase a $1,000+ FXO PCI card for the Asterisk computer and have echo. I would rather spend $1,500 to $2,000 on an entire new phone system that we can use true analog with. Although we will be missing out on some yummy Asterisk features.
Have you looked at the router that you are using ? I had a client that had an expensive vpn router that was ruining all the voip calls. I put in a basic Linksys router and it worked fine. In general I use the SMC Barricade router (I believe the model is the 7404 - or something close to that). If you do use the SMC make sure that the extra firewall settings are off. Also what speeds do you have for your DSL ?
It could be your ISP, we had the similar situation with one of the remote offices (central office as a VoIP privider). Data traffic seemed fine but voice quality wasn’t acceptable. After researching the situation we found that local ISP had some issues with traffic balansing through two different carriers. Voice packets were coming not in order through different pathes and so often codec just dropped it causing fuzzy sound. It was very hard to convince the local ISP to resolve this issue since all speed test indicated great speed and reliability.
The issue is not with the maturity of the technology. There is an issue with your broadband connection, your network, or your service provider. Have you tried Junction networks junctionnetworks.com/index.php? We have been using them for over a year now and get crystal clear calls over our cable modem. They also accept IAX trunks from their Asterisk customers.
One downside of broad band is that you cannot get QOS over your ISPs network, so there is no way to prioritize your VoIP traffic to the internet. This probably is not an issue on a smaller deployments, but you do need a reliable connection to ensure voice quality. If you don’t feel that your VoIP provider or your network is the issue and you cannot get decent broadband service in your area you may want to consider adding a Digium card to your PBX and using POTs lines.
Ditto on this one, I use sip from junction over broadband at a couple of sites. Round-trip from sites in New England is ~18.5ms. That is very good.
Try “traceroute -q 12 provider.com” where provider.com is their sip/iax server. The 12 (for example) samples time to and from each hop en-route and give a fair idea of what you might expect. test at different times of the day. A good way to identify bottlenecks on the net.
Junction also has good service - someone always answers the phone - and takes care of any question or issue.
[quote=“djmonroe1”]The issue is not with the maturity of the technology. There is an issue with your broadband connection, your network, or your service provider. Have you tried Junction networks junctionnetworks.com/index.php? We have been using them for over a year now and get crystal clear calls over our cable modem. They also accept IAX trunks from their Asterisk customers.
One downside of broad band is that you cannot get QOS over your ISPs network, so there is no way to prioritize your VoIP traffic to the internet. This probably is not an issue on a smaller deployments, but you do need a reliable connection to ensure voice quality. If you don’t feel that your VoIP provider or your network is the issue and you cannot get decent broadband service in your area you may want to consider adding a Digium card to your PBX and using POTs lines.[/quote]
We sell Digium as well as other brands, and could also help you set it up and configure it through our integration services so you don’t have an echo issue. von-supply.com/product_info. … ucts_id/66
[quote=“KuJaX”]Another alternative is getting a two or four port FXO card for the Asterisk box to convert the analog signal to digital format. Problem with that, is that I have heard from nearly every Asterisk expert I have talked to that the technology isn’t there and you’ll still get really bad echo occasionally from the analog to digital FXO conversion.
What is your opinion(s) on this matter? I don’t want to purchase a $1,000+ FXO PCI card for the Asterisk computer and have echo. I would rather spend $1,500 to $2,000 on an entire new phone system that we can use true analog with. Although we will be missing out on some yummy Asterisk features. [/quote]I’m running two, 4 port FXO Digium cards simultaneously in a TB 1.2.3 server at a client. There are 3 inputs in each in use from 2 different POTS providers. One is Verizon (Centrex) pure POTS, the other is VoIP Optimum Voice from Cablevision, who do not provide credentials and force end users to use the analog output of their ATA’s (Moto SB5150 IIRC) (their VoIP is over their private packet switched docsis subsystem).
There are 6 users in the office (Aastra 9133s and 480i CT’s) and it’s quite a busy environment with around 6000 inbound minutes per month (and somewhat less than that out). There’s really no noticable echo at all to speak of (if any at all) anymore. In the past on a AAH 2.6 with Grandstream GXP-2000’s, there was some when the handset volume was at or near max, but on this version of *, that’s gone, even on the gxp’s (which aren’t really used anymore).
I have been getting complaints every few weeks that not a day goes by without most users (if not all) experiencing some dropped calls. Not a show stopper (probably a few each day) but I hear the noise from them. Since they really can’t be bothered to “log” info when a call drops (in or out, cell or land, etc) as I’ve requested, and I cannot find any obvious log entries when grepping for “fail” or “error” (I’m really not quite sure what I should be looking for anyway), I have no idea where the issue might be.
I also have a WRT54G running HyperWRT for the LAN and I plan on adding a trunk for Vitelity very soon and testing it initially for outbound quality and reliabilty. Afterwards, I will test both out and in. I will be very curious to see if my calls over pure VoIP suffer any of the drops I’m getting complaints about. I have a sneaking suspicion that the current dropped calls have something to do with the FXO cards, perhaps something along the lines of the cards are not being sensitve enough to line voltage and sometimes think the POTS trunk has been closed when it really hasn’t-- but that could be wrong, and I don’t even know how to check for that condition.
So, if echo is your biggest gripe, I’d say you need not worry too much there. You do have good IP phones also…
