This is cool. It's nice that you can register an ATA.
This is not really new functionality though, I mean any service can do this from voip.ms to flowroute.
I think it would be cooler if the article showed how you can use the scriptability of Plivo in tandem with their SIP registration to really make a professional supplemental phone. In addition, demoing how to provision your cellphone to call forward to your landline on no-answer would be useful.
Not a bad blog post, just missing that extra sauce that would really make it pop for me.
Funny enough, I did this last month using the Telephone Mac application (just as the author initially did). The real story that the article doesn't quite do justice to is how easy it was. From zero to a functioning phone (making and receiving calls) in less than 30 mins.
If you were to recreate this via FreeSWITCH/Flowroute, it's more like an hour and change from scratch. (Register for VPS, install everything, etc.)
Also, got to give them props on support. I've stepped away from doing this for a living, but wanted an easy way to call my relatives overseas. The first attempt resulted in poor quality to the foreign landline. Cell was crystal clear. I notified Plivo team via the built-in chat and they had it fixed promptly.
This was a request from an extremely low-volume user calling European country Plivo doesn't support for anything other than dialing to. In other words, not a lot of $$$ to be made, but service was top notch!
One thing to be wary of as with most voip providers is E911. I'm not sure if Plivo offers it but it's worth looking into before venturing down this path.
From their terms of service it doesn't appear that they do offer it.
"You also acknowledge and understand that Plivo does not currently allow you to access any 911 or similar emergency services (no traditional 911, E911, or similar access to emergency services). The Services are not intended to replace any primary phone service, such as a traditional landline or mobile phone, that may be used to contact emergency services."
I also use voip.ms. Unfortunately their API and features have stagnated for quite a while. Even a basic feature like the ability to upload a voicemail greeting has not been implemented yet.
That being said the features they do have implemented work well and the pricing is attractive.
I can't really tell what Plivo offers other than an API to build custom solutions.
I use it as well.
It does seem to have slowed in terms of new features.. but it's the absolutely no-hassle, reliable, functional service that I just love. It's cheap, it does what it does properly, you don't get spammed or tricked into anything whatsoever, and if you do happen to have a billing issue or something, their support has always come through over email with no problems at all for me.
They have their act together.
Custom voicemail system with per-contact messages.
I just ported my home VoIP service from Time Warner, with their Motorola ATA, to an OBi202 with Anveo. (The OBi202 is also connected to my Google Voice account, so when I'm home where I have terrible cell service, I can still receive/make calls from my "cell" #.)
Anveo's Visual Call Flow Builder is pretty neat and can do what the OP mentions. I have a white list of #'s that will ring through to my home, a black list of #'s that have to press 1 to leave a message, and everything else just goes straight to voice mail. It looks like this - http://i.imgur.com/lEe8Nbv.png
It was all pretty straight-forward to set up and works great so far. Call quality is excellent.
OP here. That looks awesome -- exactly the kind of thing I want to set up one way or another.
I also want to have a Google Voice-style online interface for voicemail, starting conference calls, etc. I think there's a lot of potential in an open-source web app that has these features and makes the SIP setup stuff a little bit easier than having to go through the general Plivo dashboard.
For easier setups and ludicrous pricing, I can recommend DIDlogic. Supports call groups, has incoming phone numbers in many many countries and is dirt cheap. Haven't had any issues with them either.
Ugh, this is nothing new and the author is relatively uninformed on VoIP in general. SIP trunks got hot in the consumer space around 2005-2006 and there are much better alternatives than the author describes as well as far superior hardware (IMHO that Cisco ATA is junk for the price).
Do your research. Use something like the PBXIAF forums and software if you want a good place to start. If you don't need 911, which all providers have to offer (paid is the Q), leverage GVoice with an Obi (someone already mentioned) if you want to get it on the cheap but have hardware you can grow into an actual PBX.
Also, I think it's worth noting how great the Obihai is. It's a great little ATA with a ton of features.
My only complaint is that the Obi can be confusing from a consumer perspective. It's not clear, from their marketing, which services are on the Obihai and which are from the operator.
So yeah, the little Obihai is a kickass little adapter. It even does direct Obihai to Obihai mesh networking; a neat little trick.
As a newcomer to VoIP, a lot of it is confusing. I don't consider myself dumb or technically naive, and I've found it pretty difficult to get my head around how all this stuff works.
The Obi is so close to being an awesome consumer device and it was no problem for me to set up, but it's not at the "I can buy this for my parents" level of easy yet.
(FWIW, people like windexh8er totally don't help this situation. Well-reasoned suggestions and links to good documentation would serve the community better than trollery.)
If you wanna get an idea of how stuff actually works in VoIP, we cover the topics in depth every two weeks in our Expert Q&A sessions (next one is on Porting here:[0])
Here's a link[1] to 4 of the past ones so you can get an idea. We'll be publishing the videos on youtube soon. If you attend these, you'll quickly start to get a good handle on the world of communication; it's just a TON of acronyms.
Here's a link to one of my favorite blogs on VoIP [2].
I would love to see videos of these sessions! I also have a pretty technical background, but a lot of the VoIP stuff I don't have a great understanding of.
Also, Andorid now supports SIP in the native dialer client and it works over wifi and cellular data, so you don't even have to install anything extra to get it working on an android phone.
Android can support SIP protocol in the native dialer, but most phones I've looked at do not have SIP in the dialer they ship with the phone.
If you have a Nexus 4, it does support SIP, but your audio quality is usually terrible in a real world, on the street situation, where there is background noise. I have had no end of complaints, regarding the audio quality on my Nexus 4, from people on the other end of the phone call.
Finally, after much grinding and nashing of teeth, and assuming I had bad hardware, I finally found the following thread.
SIP is unusable on the Nexus 4 because the device doesn't provide an
AcousticEchoCanceler[1], AutomaticGainControl[2] or NoiseSuppressor[3].
Additionally, the VOICE_COMMUNICATION[4] audio source designed for VoIP
applications returns the same audio as if the application asked for raw
microphone data, which is completely unusable for VoIP calls.
This has been a source of great frustration to me, since I run my own FreeSWITCH installation, and wanted to use it via my Nexus 4. At this point, I am forced to go back to non-VoIP.
I've been watching that microphone issue for a while now - is there no hope of applying those quality functions to the device so that SIP usage would be bearable ?
Yes, this is true: SIP is in Android but unless you have CPU-based AEC AGC, etc., which is for the OEM to license, and nobody worked this out with LG for the Nexus 4, it will sound like crap.
I'm still waiting for 4.3 so I can't say if this has been improved.
You clearly know this stuff. You should expand this into a step-by-step article, with what to use rather than just what not to use. Denouncing the article we have without providing clear alternatives may give you and some in-the-know friends a good time, but it's useless to me. You might consider hooking the rest of us up with some actual knowledge, rather than a vague incitement to "do research" on "forums."
There are plenty of step-by-step articles out there. Personally, I use FreePBX to manage an Asterisk system with Grandstream, Cisco and Siemens endpoints. Any google search for FreePBX will find lots of guides.
If you can get your cellular provider to give you one (typically you have to be on a post-paid plan), great. Otherwise they are a few hundred dollars last time I looked, and still require you to have at least one bar somewhere in your home. I do not consistently have that from anywhere in mine, although maybe a femtocell's better antenna would.
29 comments
[ 0.22 ms ] story [ 85.0 ms ] threadThis is not really new functionality though, I mean any service can do this from voip.ms to flowroute.
I think it would be cooler if the article showed how you can use the scriptability of Plivo in tandem with their SIP registration to really make a professional supplemental phone. In addition, demoing how to provision your cellphone to call forward to your landline on no-answer would be useful.
Not a bad blog post, just missing that extra sauce that would really make it pop for me.
Source: I'm a big telecom nerd.
You are not locked into using Plivo's carriers. You can still use voip.ms with Plivo. Ping support@plivo.com and they will help you out.
If you were to recreate this via FreeSWITCH/Flowroute, it's more like an hour and change from scratch. (Register for VPS, install everything, etc.)
There's always a market for well done abstraction.
Also, got to give them props on support. I've stepped away from doing this for a living, but wanted an easy way to call my relatives overseas. The first attempt resulted in poor quality to the foreign landline. Cell was crystal clear. I notified Plivo team via the built-in chat and they had it fixed promptly.
This was a request from an extremely low-volume user calling European country Plivo doesn't support for anything other than dialing to. In other words, not a lot of $$$ to be made, but service was top notch!
From their terms of service it doesn't appear that they do offer it.
"You also acknowledge and understand that Plivo does not currently allow you to access any 911 or similar emergency services (no traditional 911, E911, or similar access to emergency services). The Services are not intended to replace any primary phone service, such as a traditional landline or mobile phone, that may be used to contact emergency services."
That being said the features they do have implemented work well and the pricing is attractive.
I can't really tell what Plivo offers other than an API to build custom solutions.
Yes, it can be quite techy, but nothing new.
Edit: Phone bill is approx $10/month with $1.50/month for E911
I just ported my home VoIP service from Time Warner, with their Motorola ATA, to an OBi202 with Anveo. (The OBi202 is also connected to my Google Voice account, so when I'm home where I have terrible cell service, I can still receive/make calls from my "cell" #.)
Anveo's Visual Call Flow Builder is pretty neat and can do what the OP mentions. I have a white list of #'s that will ring through to my home, a black list of #'s that have to press 1 to leave a message, and everything else just goes straight to voice mail. It looks like this - http://i.imgur.com/lEe8Nbv.png
It was all pretty straight-forward to set up and works great so far. Call quality is excellent.
I also want to have a Google Voice-style online interface for voicemail, starting conference calls, etc. I think there's a lot of potential in an open-source web app that has these features and makes the SIP setup stuff a little bit easier than having to go through the general Plivo dashboard.
Do your research. Use something like the PBXIAF forums and software if you want a good place to start. If you don't need 911, which all providers have to offer (paid is the Q), leverage GVoice with an Obi (someone already mentioned) if you want to get it on the cheap but have hardware you can grow into an actual PBX.
$0.02.
My only complaint is that the Obi can be confusing from a consumer perspective. It's not clear, from their marketing, which services are on the Obihai and which are from the operator.
So yeah, the little Obihai is a kickass little adapter. It even does direct Obihai to Obihai mesh networking; a neat little trick.
As a newcomer to VoIP, a lot of it is confusing. I don't consider myself dumb or technically naive, and I've found it pretty difficult to get my head around how all this stuff works.
The Obi is so close to being an awesome consumer device and it was no problem for me to set up, but it's not at the "I can buy this for my parents" level of easy yet.
(FWIW, people like windexh8er totally don't help this situation. Well-reasoned suggestions and links to good documentation would serve the community better than trollery.)
If you wanna get an idea of how stuff actually works in VoIP, we cover the topics in depth every two weeks in our Expert Q&A sessions (next one is on Porting here:[0])
Here's a link[1] to 4 of the past ones so you can get an idea. We'll be publishing the videos on youtube soon. If you attend these, you'll quickly start to get a good handle on the world of communication; it's just a TON of acronyms.
Here's a link to one of my favorite blogs on VoIP [2].
Hope that helps!!!
[0] http://www.eventbrite.com/myevent?eid=7476736119
[1] http://blog.2600hz.com/post/52151527067/2600hz-expert-q-a-re...
[2] http://200ok.info/
http://www.obihai.com/googlevoice.php
$40 - $70 on Amazon depending on model / features.
http://www.amzn.com/B007D930YO
If you have a Nexus 4, it does support SIP, but your audio quality is usually terrible in a real world, on the street situation, where there is background noise. I have had no end of complaints, regarding the audio quality on my Nexus 4, from people on the other end of the phone call.
Finally, after much grinding and nashing of teeth, and assuming I had bad hardware, I finally found the following thread.
http://forum.xda-developers.com/showthread.php?t=2228361
Quoting from the thread
This has been a source of great frustration to me, since I run my own FreeSWITCH installation, and wanted to use it via my Nexus 4. At this point, I am forced to go back to non-VoIP.I'm still waiting for 4.3 so I can't say if this has been improved.