Hosting Providers, # of POTS lines, and voicemails as MP3's

Asterisk Asterisk Support
1

Hi, I’m setting up a small business where I’ll be having people in the field dialing into an asterisk server via a toll-free number and leaving voicemail messages. I have three questions:

  1. I’d rather have the phone line(s) be POTS vs. VOIP because I want the recording quality to be as pristine as possible. I’ll have at most 5 concurrent callers, how many POTS phone lines would I need?

  2. Because I’d want folks in the field to access and play the voicemails over the web, bandwidth is fairly important, and again it might be difficult for me to get good bandwidth via a cable modem or DSL line from home. Is there a list of companies that will host an asterisk server where I can get shell access, and where they have the facilities to run some 1-800 lines into them?

  3. Is there an easy way for me in asterisk to have people download their voicemail messages as MP3 files?

Thanks in advance.

2

OK I know where you can do the Hosting in the USA (MoJoMonster.com) I have used them for about 6 years now.

Chad can customize the Asterisk server if needed, They do custom Asp / Php webbased apps.

They can also get your setup with a full Asterisk server complete setup with ARI / Astedrix / click to call stuff

ARI littlejohnconsulting.com/

It is a web interface for Voice Mail and recorded calls,
you can not beat it FREE listen to and download as wav

AsteriDex a web-based AutoDialer
nerdvittles.com/index.php?p=83

As for POTS Vs VoIP inbound (in the US using someone like Connect.VoicePulse.com) you can get mutli- channels for inbound calls and increase / decrease as needed via thier website.

I do not think you would notice the VoIP or POTS on a good hosted server sitting on the backbone with a soild 100 meg link and no lag time makes for a SWEET sound you can use the Ulaw as bandwidth is not a issue

3

[quote=“aaelghat”]Hi, I’m setting up a small business where I’ll be having people in the field dialing into an asterisk server via a toll-free number and leaving voicemail messages. I have three questions:

  1. I’d rather have the phone line(s) be POTS vs. VOIP because I want the recording quality to be as pristine as possible. I’ll have at most 5 concurrent callers, how many POTS phone lines would I need?
    [/quote]

Depends on how many concurent calls you need at a time. If everyone will be calling in at once then you need all of them. You can get a few pots lines and if they are busy have them forward to VOIP.

Although cable is good and I have clients that use it, it isnt the best. If your cable is down you are in trouble. On the other hand data centers charge you for bandwith while cable vision dosent. I have multiple serevrs in dedicated centers that are on pure VOIP and I have had no problems since.

[quote=“aaelghat”]
3) Is there an easy way for me in asterisk to have people download their voicemail messages as MP3 files?
[/quoute]

Yes. I know that asterisk emails them as wav. You can create a script that converts all of your Wav. Files in to MP#'s.
Thanks in advance.[/quote]

4

What you can do is have the voicemail recorded in wav format and have it sent to their emails, which * does as soon as the voicemail is completely recorded. So they won’t even have to call in to listen to it. But if bandwidth and call traffic is what you are concerned about and don’t want to spend so much, you can create redundancy by getting a Cable internet connection, a DSL connection, and if you can get Verizon FiOS broadband connection altogether into a multi-uplink router such as the Linksys RV016 VPN Router [up to seven (of the 16) port internet connection failover or load balanced redundancy] then you should be fine as far as bandwidth, redundancy and cost are concerned at about $120/month. As for telephony, I second Dovid, you should get both POTS and VoIP lines for several reasons, clarity is not one of them because you can hardly tell the difference between a voicemail on either as long as they are wav format. Besides that you might want to send calls to the VoIP lines when the POTS are busy, you also want your long distance or all non-local calls to use the VoIP lines primarily and perhaps use POTS when all VoIP lines are busy to cut cost. i hope I’m making sense.

5

Thanks so much for the responses so far!! OK, you’ve convinced me that if I get VOIP lines and my asterisk server is located on a backbone, than I don’t have to worry about call quality and I can ditch the POTS lines. A bit more information, and a few more questions…

Basically, * will only be receiving incoming calls, and I’d like to give each person their own voicemail box (which I want to password protect so a passcode needs to be entered before they can leave the voicemail). I don’t want to give an IVR message like (for Bob, press 1, for Rachel, press 2, because I don’t want callers to know who else has a voicemail box). I guess I could set-up long extensions, and the “passcodes” that people are prompted for are really the extensions. The whole purpose of the asterisk box is to allow people to call in and leave confidential messages, so I just need to worry about inbound calling, and not outbound. I want each person to have their own toll-free number, and I went to the connect.voicepulse site. It looks good, but each toll free line is $11 month, I’ll keep researching, but let me know if there is something more cost effective.

I want to be able to batch download the voicemails and burn them to a CD which I would send back to the caller.

Thanks for the info on ARI - I have Trixbox 1.1 which is bundled with ARI, but after doing some more research I also found this.

serge.mankovski.com/voip/asteris … -generator

Which does the WAV to MP3 conversion, and gets me closer to automating the burning to CD process.

Also, thanks for the info on MoJoMonster. I’ll call them on Monday, are their other hosting services that will let me host an * box?

6

This sounds like one of those dating services where people call in and leave messages for whoever they are seeing. There is a way to do this, what you need to do is when the calls come in you can implement a dialplan that will have an option such as ‘to leave a message for your party, press 1’. So when they press 1 it takes them to a dialplan that says something like ‘please enter your parties 4-digit code’, so when they enter this number * dials the number but since there is no phone configured for the number it will go straight to the number’s voicemail. The next thing you will have to do is to modify the unavailable voicemail IVR to say something like ‘after the tone please leave a message for this person’ or even batter if you can find a way for * to replace “this person” with the person’s name.

extensions.conf

[sip.inbound.com] ;This is going to be your inbound trunk, telling all incoming calls where to go, whether to go to the same or diferent dialplans

exten=>18001234567,1,Goto(inbound|s|1)
exten=>18007654321,1,Goto(inbound|s|1)

[inbound]
exten=>s,1,Ringing
exten=>s,2,Answer
exten=>s,3,Wait,2
exten=>s,4,SetMusicOnHold(default)
exten=>s,5,Background(welcome to …, to leave a message for your party press 1, etc)
exten=>s,6,WaitExten(10)
exten=>s,7,Background(r-u-still-there…,to-leave-a-message-for-your-party-press-1, etc)
exten=>s,8,WaitExten(10)
exten=>s,9,Hangup

exten=>1,1,Goto(inbound-client|s|1)

[inbound-client]
include=>4-digit-numbers
exten=>s,1,DigitTimeout(10)
exten=>s,2,Set(TIMEOUT(absolute)=240)
exten=>s,3,Wait(1)
exten=>s,4,SetMusicOnHold(default)
exten=>s,5,Background(please-enter-your-partys-4-digit-number)
exten=>s,6,Read(extension,4)
exten=>s,7,Dial(SIP/${extension},30,m)
exten=>s,8,Voicemail,u${extension}@default
exten=>s,9,Playback(invalid)
exten=>s,10,Goto,s|3
exten=>s,11,Background(r-u-still-there-plz-enter-partys-number)
exten=>s,12,Read(extension,4)
exten=>s,13,Dial(SIP/${extension},30,m)
exten=>s,14,Voicemail,u${extension}@default
exten=>s,15,Playback(invalid)
exten=>s,14,Playback(goodbye)
exten=>s,15,Hangup

[4-digit-numbers] ;All 4-digit numbers entered by inbound callers must be configured in sip.conf to register to this context
exten=>s,1,Dial(SIP/${extension},30,m)
exten=>s,2,Voicemail,u${extension}@default

I hope this helps.