[WM: Seriously, you’re commenting on an article that was written almost 8 years ago!! Try something a bit more recent like this one.]

[WM: Not surprisingly, these disputes usually turn into "he said, she said" shouting matches. As we have cautioned on many occasions, the consumer is best served by having multiple VoIP providers. Separating out your inbound DID traffic from your outbound calling is also a good idea especially where high call volumes (either in minutes or number of calls) typically would flag your account as suspect. Anyone with teenage (or older) children probably would not consider the call volumes in question here surprising. However, fraud on residential VoIP lines is rampant. That’s why we continue to recommend to VoIP providers that they specify usage caps so that consumers know up front what the ground rules are and don’t get blindsided. Some providers do it. Most still don’t.]

[WM: TelaSIP’s response follows.]

This is the response we sent to this customer. Unfortunately, he did not contact us prior to the posting. This is a bad time of year for fraud so we had to turn up our account audits. We did send him an email requesting information about the usage on the account. He did not reply, but decided to complain. Our experience is that customers that don’t deny they are using the service for business usually are.

We do not have a cap on our service. We have customers that use the phone more than what you describe below. Unfortunately, we have too many customers that sign up for residential service and use it for business. It would be nice if we could just assume that everyone is being honest. The incidence of fraud is increasing as well so we need to be able to audit accounts and disable them in order to stay in business. It was not the number of minutes that raised the flag, but the number of calls. You placed 350 calls in a two week period. Even with 6 people using the phone this is unusually high.

We understand your disappointment, but hope you understand ours. Had you contacted us and explained that you were not using the service fraudulently, we could have resolved the issue. We will set the account up as you requested below and will understand if you decide to port your numbers back to Vonage.

I have to say, though, the lack of a web interface to keep track of calls is beginning to get a bit annoying after I’ve been using the service for a few months. The quality is excellent, however, and the reliability is by far the best of any VoIP service I’ve used by a long long shot – I don’t think I’ve had a single outage in the six months I’ve been with them (that I’ve noticed) – and you can’t fault the price.

[WM: Call them if you need a specific number. They’ve got almost every NPA/NXX in the U.S.]

Ah, not so attractive lately considering TelaSIP only allow one concurrent call per trunk! Though I will agree that the quailty of service regarding the trunk is excellent, I’m not as much a fan of their pricing model. I was looking for the ability to make 4 calls simultaneously and would need to shell out for 4 trunks if I wanted that luxury.

I know a good many of the pre-existing TelaSIP customers have no such restrictions and that’s great. This note is just a word to the new customers who are considering using them as a provider. If you want just one trunk, it would be hard to beat them. If you want to utilise your Asterisk box and save money, I would think very carefully about who you choose. Just my 10 cents…

[WM: I don’t know of any all-you-can-eat provider that lets you make multiple simultaneous calls. If you want to do that, you’ll need to sign up with a provider that charges by the minute. If you need multiple simultaneous calls with an all-you-can-eat provider, then you simply buy multiple trunks. TelaSIP, like every other provider, pays by the trunk, and 1 Trunk=1 Call.]

My router is a WRT54G. Any chance you can go over some of the issues with SIP and NAT along with possible solutions?

[WM: Assuming you’ve followed through the TelaSIP tutorial and still can’t get registered, you need to contact them to get it resolved. We do intend to cover some NAT issues in a future article, but this doesn’t sound like your problem.]

How do we setup incoming call using goiax. I have followed your method. I do receive call from iax.cc and fwd

Thanks for the great articles and help.

[WM: DID’s don’t appear to be working quite right yet. Alos GoIAX is having to wrestle with security issues at the moment so their service is locked down except for 800 calls. Kepp checking their site for more info.]

[WM: I assume you’re talking about phone service for a business. If one of the prepackaged plans meets your business requirements, you are quite correct that it would be a waste of time to set up your own Asterisk server. For many small businesses, you will come out better doing it yourself. TelaSIP also makes special arrangements depending upon the size of your operation and the amount of phone use involved. For example, for doctor’s offices, I’ve heard of them offering a plan which is quite similar to their residential service except with one port per DID. That means you could have two simultaneous incoming or outgoing calls (i.e. out of the office) at the same time for about $30 a month. Four simultaneous trunks would be $60 which is about what BellSouth charges for ONE business line with NO long distance service included. TelaSIP’s plans all have free long distance throughout the continental U.S. If all else fails, drop them a note or call and ask them what they can do. It can’t hurt. Hope this helps. ]

Here you say use SIP because there’s less overhead: "Why SIP and not IAX? The simple answer is there?¢ÀÜ?°?Ǭ¢?¢‚Ǩ?°?ɂİ?Ǭ®?¢‚Ǩ?°?É‚Äò?Ǭ¢s less call overhead between you and the provider."

Can you elaborate?

Great series of articles. I’m trying to get up to speed on Asterisk and they are really helping. Thanks.

[WM: Great question, and thanks for pointing out the sloppy language. I’ve worked on it a little more. The beauty of IAX is that it reduces issues regarding network address translation (NAT) and firewall problems which tend to cause VoIP communication problems such as lack of audio in one direction or another. The curse of IAX is that it accomplishes this feat by pushing everything (signalling and voice data) through your Asterisk server using one UDP port. While you only have one port to worry about with your firewall, it also means all the work of a call has to be handled by the Asterisk server and all of the voice data has to travel in and out of the server. So there’s more call processing required with IAX, and you also have to move the entire voice call in both directions through your server so there’s bandwidth to worry about as well. With SIP, only the signalling is managed by the Asterisk server. The voice data itself travels directly between the endpoints of the call. So SIP’s more efficient once you get a call completed successfully, but getting to that point can be a bear. In short, there are tradeoffs with both protocols. Some vendors tend to be better at managing IAX while others do a great job with SIP. For endpoint connections, i.e. remote telephones, IAX is the hands down winner in my book because of the ease of configuring the phone and elimination of NAT and firewall worries. We’re going to cover Digium’s IAXy device in an upcoming article. It gives you an IAX connection to the Mother Ship from anywhere. Then all you have to do is plug a regular POTS phone into the IAXy, and you’re in business. Hope this helps.]

Odd issue with my set-up maybe you have insight into.

When dialing out on a voip trunk, the ring back tones we hear are normal: a 5-10 of silence while the call sets-up, then ringing until the caller answers on the other end.

When dialing out through the spa-3000 on the PSTN line, we hear one ring (comes from asterisk’s ring back tones to our ears, probably until the spa3000 "answers" the sip set-up from asteris, then a loong 10-15 seconds of silence (assumed to be while the spa3000 dials the call), then normal ringing until the other end answers. The long silence confuses some of our folks.

Any idea how to mitigate that 10-15 seconds of silence?

[WM: Try putting the following entry in the [general] section of your sip.conf file: progressinband=yes]

I have to admit that after reading the feedback comment for last weeks article I signed up for Telasip yesterday, drat…. (I did send Gene an email after I read today’s article giving your blog credit). I look forward to trying out his service.

Keep up the good work.

[WM: If you’re talking about residential service with TelaSIP, the answer is 2. I’m not sure why most residential customers would need much more than that.]

Will there be a future article on nat traversal for Asterisk? I’m trying to set up my unlocked [;)] Linksys PAP2 to connect to my asterisk server at work from my isp at home, with so far very little luck. Your articles have been quite useful thus far.

Keep it up eh!