Ask me anything about SIP and VoIP Telephony
Frequently Asked Questions
1) What are some recommended VoIP providers?
- BabyTel (YMMV, I do not personally recommend them anymore. They are very unreliable.)
- Les.net (wholesale DID and termination)
Feel free to recommend some more!
2) What about 911 service? Does it work?
When you are looking for a VoIP provider, one thing to keep in mind is 911 service. If you really need a provider with 911 service, look at their website or contact them to inquire about whether it is available. Not all of them will provide the service but many major Canadian VoIP companies will provide what's called "e911 service" You should note however, the reliability of such a service is questionable and it is recommended you keep a copper line connected with a spare analog telephone, you do not need an active telephone service to dial 911.
3) Can i port my number to a VoIP provider?
Yes. Some VoIP providers offer to port your number from a landline or cell. You should check with the provider whether they offer porting and your number can be ported. Not all exchange numbers can be ported depending on location and whether the VoIP provider is able to port numbers. The porting process can take about 2-3 weeks, unfortunately because of the way porting works this is the minimum time required to port a number.
4) I heard about this thing called Asterisk, what is it?
Asterisk is a linux based PBX application. You can use it to route calls over IP and take PSTN calls and convert them to VoIP. You can also do more advanced functions such as being able to setup a service to call in from any number and have it route long distance calls so the receiving telephone number is not charged for the call. If you are curious about Asterisk and want to learn how to set it up, visit the following site: http://www.asteriskguru.com/
NOTE: If your question is not covered in this FAQ, feel free to post!
Tips and Tricks
1) Styles posted the following to do with audio quality and Linksys ATA devices:
I've noticed that quite a few people are using Linksys devices. The Linksys devices seem to come with a default RTP Packet Size setting of 0.030. In my case, I have found that call quality is sub par with this setting. Changing this to 0.020 has improved the call quality for me, especially with inbound calls from les.net.
This setting was recommended to me by les.net support. They mentioned many people in the Toronto area have had call quality issues that were resolved by changing this setting. It is also recommended for use with FreeSWITCH. See here for more info: