-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I've just ported several numbers to Voip.ms, and I continue to be amazed at the features and flexibility built right into their Web interface. I'll be building a local phone switch server (Asterisk or similar) sometime in the future, but for now Voip.ms provides all the tools I require. I've used a Cisco SPA-122 ATA and a Grandstream HT702 ATA, as well as a Grandstream GXP280 voip phone (and a number of SIP phone applications on computers and Android phones). All configure Caller ID a bit differently, but I've been able to set up different Caller IDs for each phone, even when they dial out from the same number. One account has two different DID (Dial-In) numbers. A single phone on that account is configured to use one number for its caller ID, but receives calls from both DID numbers. It's a bit of a pain to switch the caller ID to the other number, and I don't think it's really necessary; as long as that phone can receive calls from both numbers it doesn't matter what number it displays on outgoing calls. YMMV. - --Bob. On 31/12/15 10:50 PM, Blaise Alleyne wrote:
On 31/12/15 05:42 PM, David Mason wrote:
I know there are some people on here that know about VOIP and Asterisk, and I'm looking for some pointers. I've looked at Asterisk a bit (a while ago) and it was kind of overwhelming.
I'm running a FreeSWITCH server at work. More overwhelming in some ways, less in others. There's just a lot of telephony stuff you need to get up to speed with...
Though, at home, I'm just using SIP clients, connecting directly to VoIP.ms SIP servers in Toronto and using their web portal to configure call forwarding, IVRs, etc.
I want to have several phones and several phone numbers (a 416 for where I work, a 905 for where I live, and a 902 for where a bunch of my friends and family are).
VoIP.ms
I want to pick up a phone, dial a number and have the appropriate VOIP phone number be used.
Caller ID is a little trickier, but definitely doable. With SIP and VoIP.ms, there are two options:
Option 1: Set the Caller ID in your VoIP.ms account profile. This would always be the same, wouldn't be able to use different numbers for outgoing caller ID. This is what I do at home.
Option 2: Set VoIP.ms to accept the caller ID from your calls. With my FreeSWITCH server at the office, we have it set to use different outbound caller IDs depending on a prefix dialed before the number. Maybe there's a way to set it in your SIP client without needed an Asterisk/FreeSWITCH server in between you and VoIP.ms? You'd need to find a way to set this and trigger the appropriate caller ID on your end though before it hits VoIP.ms, if you want it to vary.
I'd like a different ring tone from each of the numbers.
Also a little tricker, but probably doable. The ring tone is going to be set by your SIP client, whether it's a softphone or handset.
(a) You'd need a SIP client that can vary the ringtone based on the incoming call.
(b) You'd need a way to identify the number used by the incoming caller.
So, if John Doe calls you on your 902 number, your SIP client won't know that it was on the 902 number -- it will just see John Doe's name and number.
However, VoIP.ms has an option for DIDs (Direct Inward Dialing numbers, i.e. traditional phone numbers) to prefix the caller ID. So, at the office, we've got caller ID prefixes. If John Doe called Alleyne Inc., the caller ID gets modified by VoIP.ms so that the name is "[AI] John Doe <X>" (where X is John's number). The "[AI] " part tells us visually in the caller ID what line he called into.
Presumably, you could use that information to select a ring tone as well, but I'm not sure that'd be easy to do on your average SIP client. If you're going through Asterisk/FreeSWITCH, you'd have more opportunity to mess with the Caller ID for sure. I'd imagine some SIP clients could have different ring tones for different caller IDs, but not sure how flexible.
I'd like an answering machine function with multiple mailboxes for various family members.
The VoIP.ms service does this easily with it's subaccounts. You can also definitely do this with an Asterisk or FreeSWITCH server of your own.
I'd like to be able to access those messages (or at least be able to see that there are pending messages) from an computer.
With VoIP.ms at home and FreeSWITCH at work, I get copies of voicemail messages by email, with a WAV or AIFF file attached. That makes it easy to listen to messages.
To manage (i.e. delete) messages, I still dial in to a voice menu. There was a web interface I was playing around with for FreeSWITCH, but didn't spend enough time to get it working.
Some of these may not be possible/easy, but that's what I'd *like*.
The cost/benefit I'd recommend is to try VoIP.ms, and see how far you can get with just their services and SIP clients. You'd definitely have more power and flexibility with your own Asterisk/FreeSWITCH server, but it takes a lot more work to understand, setup, and manage.
I'm happy with just VoIP.ms for family stuff. To satisfy requirements at work, FreeSWITCH made more sense.
I currently have the 416 number via Primus which connects to a VOIP box (a Cisco SPA-122 that Primus provided).
You can port existing telephone numbers to VoIP.ms like you would any other provider. I've ported several Bell numbers to VoIP.ms.
Any recommendation for a cheaper/better supplier for the multiple numbers I want to have?
I'd definitely recommend VoIP.ms.
I'd heard good things about another provider back cira 2010/2011 when I was looking... Callcentric I think? I've never used them though. VoIP.ms has worked well -- very inexpensive, and with SIP, fully interoperable with free software options. For technical users though.
Bob Jonkman <bjonkman@sobac.com> Phone: +1-519-635-9413 SOBAC Microcomputer Services http://sobac.com/sobac/ Software --- Office & Business Automation --- Consulting GnuPG Fngrprnt:04F7 742B 8F54 C40A E115 26C2 B912 89B0 D2CC E5EA -----BEGIN PGP SIGNATURE----- Version: GnuPG v2.0.20 (GNU/Linux) Comment: Ensure confidentiality, authenticity, non-repudiability iEYEARECAAYFAlaHgkEACgkQuRKJsNLM5erzZACffNEIDwXsIYwNqrQNCc1LrQo0 upAAoIKRAu8JexzgWO2jPQEXQsql+byR =Srrk -----END PGP SIGNATURE-----