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 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). I want to pick up a phone, dial a number and have the appropriate VOIP phone number be used. I'd like a different ring tone from each of the numbers. I'd like an answering machine function with multiple mailboxes for various family members. 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. Some of these may not be possible/easy, but that's what I'd *like*. I currently have the 416 number via Primus which connects to a VOIP box (a Cisco SPA-122 that Primus provided). Any recommendation for a cheaper/better supplier for the multiple numbers I want to have? Thanks for any help! ../Dave
Take a look at voip.ms (http://voip.ms). They are a global voip wholesaler based in Montreal. I think you should be able to accomplish much of what you list using their service. They are certainly inexpensive. On Thu, Dec 31, 2015 at 5:42 PM, David Mason <dmason@ryerson.ca> 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 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). I want to pick up a phone, dial a number and have the appropriate VOIP phone number be used. I'd like a different ring tone from each of the numbers. I'd like an answering machine function with multiple mailboxes for various family members. 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. Some of these may not be possible/easy, but that's what I'd *like*.
I currently have the 416 number via Primus which connects to a VOIP box (a Cisco SPA-122 that Primus provided). Any recommendation for a cheaper/better supplier for the multiple numbers I want to have?
Thanks for any help! ../Dave
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The one I have used is http://voip.ms Pretty Darn cheap On Dec 31, 2015 5:42 PM, "David Mason" <dmason@ryerson.ca> 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 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). I want to pick up a phone, dial a number and have the appropriate VOIP phone number be used. I'd like a different ring tone from each of the numbers. I'd like an answering machine function with multiple mailboxes for various family members. 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. Some of these may not be possible/easy, but that's what I'd *like*.
I currently have the 416 number via Primus which connects to a VOIP box (a Cisco SPA-122 that Primus provided). Any recommendation for a cheaper/better supplier for the multiple numbers I want to have?
Thanks for any help! ../Dave
--- Talk Mailing List talk@gtalug.org http://gtalug.org/mailman/listinfo/talk
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.
On 31/12/15 10:50 PM, Blaise Alleyne wrote:
[...] 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 should add for clarity that I meant your own Asterisk/FreeSWITCH server *in between*. VoIP.ms still makes sense regardless of whether you're connecting to it with SIP clients directly, or connecting to it with an Asterisk/FreeSWITCH server and connecting your SIP clients to Asterisk/FreeSWITCH. The latter option is more work but gives you more power; trying the former first is a good intermediate step that can satisfy your simpler requirements fast and defer the decision on whether or not to host your own server.
-----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-----
On 02/01/16 02:54 AM, Bob Jonkman wrote:
[...] 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). [...]
Hey Bob, any recommendations for SIP clients? Do you like the GXP device? And which Android SIP client do you prefer? In terms of handsets, I'm managing some Aastra 9143i handsets, which were great for a few years but then mysteriously refuse to boot for a few weeks at a time. I've been looking at Granstream GXP or Polycom for replacements... On the Android side, my wife has been using CSipSimple, but I'm not sure that's the best option. I'm getting an Android device in a few days and will begin playing around with it. (On my N900, Maemo Fremantle has integrated SIP support with the Phone application, and I've mostly used Ekiga as a desktop SIP client.)
-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 On 02/01/16 11:40 AM, Blaise Alleyne wrote:
Hey Bob, any recommendations for SIP clients?
On the desktop (Linux Mint Debian Edition I) I initially had the most success with Jitsi. After some fiddling to get Ekiga to work it has become my preferred client. I don't use it much, I'd rather talk on a real handset. I have Linphone installed too, but I'm not sure it's working.
Do you like the GXP device?
The GPX280? Short answer: No. The UI on the phone is too limited to easily get last-caller info, or add entries into the phonebook (yes, I should be using an LDAP directory). The WebUI looks like it was made to be compatible with Netscape 3. The caller log info is not accessible through the WebUI. I'm sure these things can be fixed with a firmware upgrade, but this phone is no longer sold or being developed by Grandstream. But it works, and the sound quality is OK. For the next installation I'm thinking of getting a cordless WiFi enabled voip phone, something like http://en.grandstream.net/en/ip-phones/grandstream-dp715-dect-system-base-ha...
And which Android SIP client do you prefer?
I was always big on the Linphone app because of its cross-platform availability. Now that I've had a chance to use it cross-platform I'm not very impressed at all. F'rinstance, the Android Linphone does not have letters on the dialling keypad. Makes it too difficult for me to type in Voice Mail passwords. I'm now using the integrated "Internet Phone" capability in Cyanogenmod 7.2 on my LG P999 phone. That works well, but I can tell it drains the battery faster (the battery is already near end-of-life; that phone is almost five years old and not supported by anybody). I just installed CSipSimple, looks nice and clean. Something I haven't been able to do: Call one phone from another using voip.ms. Each phone (analog, VOIPphone or SIP client) has its own sub-account on voip.ms, and I've assigned "Internal extensions" to each. But I haven't figured out how to call an internal extension. This wasn't available with the POTS we had before, so it's not a loss in functionality, but it would be nice to get that working. Any hints? - --Bob. 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 On 02/01/16 11:40 AM, Blaise Alleyne wrote:
On 02/01/16 02:54 AM, Bob Jonkman wrote:
[...] 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). [...]
Hey Bob, any recommendations for SIP clients? Do you like the GXP device? And which Android SIP client do you prefer?
In terms of handsets, I'm managing some Aastra 9143i handsets, which were great for a few years but then mysteriously refuse to boot for a few weeks at a time. I've been looking at Granstream GXP or Polycom for replacements...
On the Android side, my wife has been using CSipSimple, but I'm not sure that's the best option. I'm getting an Android device in a few days and will begin playing around with it. (On my N900, Maemo Fremantle has integrated SIP support with the Phone application, and I've mostly used Ekiga as a desktop SIP client.) --- Talk Mailing List talk@gtalug.org http://gtalug.org/mailman/listinfo/talk
-----BEGIN PGP SIGNATURE----- Version: GnuPG v2.0.20 (GNU/Linux) Comment: Ensure confidentiality, authenticity, non-repudiability iEYEARECAAYFAlaIPggACgkQuRKJsNLM5epJBACeLlQ3ZKQ6D2iQpn6VK9uOIPkt ZEUAn3Ajy64wvW8a9Bneu4owt5Vm6XfA =JTNE -----END PGP SIGNATURE-----
On 02/01/16 04:15 PM, Bob Jonkman wrote:
[...] Something I haven't been able to do: Call one phone from another using voip.ms. Each phone (analog, VOIPphone or SIP client) has its own sub-account on voip.ms, and I've assigned "Internal extensions" to each. But I haven't figured out how to call an internal extension. This wasn't available with the POTS we had before, so it's not a loss in functionality, but it would be nice to get that working. Any hints? [...]
I'm not sure about VoIP.ms, with on my FreeSWITCH setup, you can just dial the username to dial internally. For example, dialing 'balleyne' or '223' on another SIP client connected to our FreeSWITCH server would ring my extension. (That is, dialing 'balleyne@thesipserver.domain.tld') I imagine with VoIP.ms, it'd be similar, with the subaccounts as usernames? That's be my first guess. Looks like there's some documentation around getting an external SIP URI for subaccounts on the VoIP.ms wiki: http://wiki.voip.ms/article/SIP_URI#Using_your_sub_account_internal_extensio... There's probably details somewhere on the wiki about making an internal call, if their system is set up for that.
On 2 January 2016 at 16:15, Bob Jonkman <bjonkman@sobac.com> wrote:
Something I haven't been able to do: Call one phone from another using voip.ms. Each phone (analog, VOIPphone or SIP client) has its own sub-account on voip.ms, and I've assigned "Internal extensions" to each. But I haven't figured out how to call an internal extension. This wasn't available with the POTS we had before, so it's not a loss in functionality, but it would be nice to get that working. Any hints?
I have this working. You just dial the extension you set up. All the extensions start with "10" and then can be whatever you like after that. So, I just dial "105" and the person with that extension set up gets called. The restriction is that both you and the person you're contacting have to be on the same VOIP.ms server. Any way, I discovered that voip.ms says you aren't supposed to be charged for voip.ms to voip.ms calls even if you use the external phone number. (though, I haven't tested it out)
-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Bob Jonkman wrote:
Something I haven't been able to do: Call one phone from another
Tim Tisdall wrote:
You just dial the extension you set up
Well, that was easy. Just surprised the heck out of my wife, who didn't believe the caller ID she saw on the phone... But the call from downstairs in Elmira to upstairs in Elmira gets routed through Toronto. That's kinda strange (and unwanted), but I'll fix that when I install an Asterisk or equivalent. Thanx, Tim!
Any way, I discovered that voip.ms says you aren't supposed to be charged for voip.ms to voip.ms calls even if you use the external phone number. (though, I haven't tested it out)
Yup, works for me. I can make calls using the regular DID phone number for the various voip.ms accounts I manage, and the charge is always 0. That was an unexpected bonus! Does anyone know if the Toronto Asterisk Users Group is still meeting? http://taug.ca/ seems to be down... - --Bob. On 04/01/16 02:31 PM, Tim Tisdall wrote:
On 2 January 2016 at 16:15, Bob Jonkman <bjonkman@sobac.com> wrote:
Something I haven't been able to do: Call one phone from another using voip.ms. Each phone (analog, VOIPphone or SIP client) has its own sub-account on voip.ms, and I've assigned "Internal extensions" to each. But I haven't figured out how to call an internal extension. This wasn't available with the POTS we had before, so it's not a loss in functionality, but it would be nice to get that working. Any hints?
I have this working. You just dial the extension you set up. All the extensions start with "10" and then can be whatever you like after that. So, I just dial "105" and the person with that extension set up gets called. The restriction is that both you and the person you're contacting have to be on the same VOIP.ms server.
Any way, I discovered that voip.ms says you aren't supposed to be charged for voip.ms to voip.ms calls even if you use the external phone number. (though, I haven't tested it out) --- Talk Mailing List talk@gtalug.org http://gtalug.org/mailman/listinfo/talk
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participants (6)
-
Blaise Alleyne -
Bob Jonkman -
David Mason -
Digiital aka David -
Nigel Auger -
Tim Tisdall