I'm looking at the http://www.grandstream.com/y-286.htm, which is an Analog Telephone Adapter for VOIP.
The ATA will initially configure its own IP address with DHCP?
'IP Address: dynamically assigned via DHCP (default) or PPPoE (will attempt PPPoE if DHCP fails and following is non-blank)' http://www.grandstream.com/user_manuals/HandyTone.pdf
Sounds like a piece of cake. Once it's configured, how is the ATA made to work with http://www.gizmoproject.com/? I would just pick up the phone and dial? There must be some configuration beyond that.
-Thufir
On Tue, 2006-10-31 at 23:45 +0000, Thufir wrote:
I'm looking at the http://www.grandstream.com/y-286.htm, which is an Analog Telephone Adapter for VOIP.
The ATA will initially configure its own IP address with DHCP?
'IP Address: dynamically assigned via DHCP (default) or PPPoE (will attempt PPPoE if DHCP fails and following is non-blank)' http://www.grandstream.com/user_manuals/HandyTone.pdf
Sounds like a piece of cake. Once it's configured, how is the ATA made to work with http://www.gizmoproject.com/? I would just pick up the phone and dial? There must be some configuration beyond that.
What does this have to do with Fedora?
In the user manual for the adapter (in the "Advanced Configuration" section), though, I note there is a configuration screen where you enter the SIP server, outbound proxy, and SIP user ID and password. Did you try reading the manual?
On Tue, 2006-10-31 at 23:45 +0000, Thufir wrote:
I'm looking at the http://www.grandstream.com/y-286.htm, which is an Analog Telephone Adapter for VOIP.
The ATA will initially configure its own IP address with DHCP?
'IP Address: dynamically assigned via DHCP (default) or PPPoE (will attempt PPPoE if DHCP fails and following is non-blank)' http://www.grandstream.com/user_manuals/HandyTone.pdf
That device looks like it sits between your phone and a network connection to the internet. If you have a modem/router you don't even need a computer. It's a stand-alone device, a network device in itself.
If you don't have a modem/router, then you'd need to use a computer to share the internet connection between itself and the ATA, much the same way as you share the internet between more than one PC.
Sounds like a piece of cake. Once it's configured, how is the ATA made to work with http://www.gizmoproject.com/? I would just pick up the phone and dial? There must be some configuration beyond that.
That's a software phone. You use your computer for VOIP, with whatever sound devices are plugged into it (mic, speakers, headphones, etc.).
On Wed, 01 Nov 2006 19:00:39 +1030, Tim wrote: [...]
That device looks like it sits between your phone and a network connection to the internet. If you have a modem/router you don't even need a computer. It's a stand-alone device, a network device in itself.
If you don't have a modem/router, then you'd need to use a computer to share the internet connection between itself and the ATA, much the same way as you share the internet between more than one PC.
I connect to a SMC7004VWBR router/firewall wirelessly, but well enough for skype. I would be using my computer as a gateway, of sorts, with this device?
[...]
That's a software phone. You use your computer for VOIP, with whatever sound devices are plugged into it (mic, speakers, headphones, etc.).
[...]
The device wouldn't work with Gizmo, then? I'd need a VoIP provider, along the lines of vonage?
thanks,
Thufir
On 11/1/06, Thufir hawat.thufir@gmail.com wrote:
The device wouldn't work with Gizmo, then? I'd need a VoIP provider, along the lines of vonage?
You definitely *don't* need Vonage. They are a walled garden ... something the Internet needs less of.
Gizmo provides a softphone but they are also a SIP provider that works with many SIP adapters. You should check their website and forums as to what info you have to plug in to your particular SIP adapter to make it work.
There are other free SIP providers too like Free World Dialup.
/Mike
Thufir:
Tim:
That device looks like it sits between your phone and a network connection to the internet.
Addendum: It *is* one of those devices.
If you have a modem/router you don't even need a computer. It's a stand-alone device, a network device in itself.
If you don't have a modem/router, then you'd need to use a computer to share the internet connection between itself and the ATA, much the same way as you share the internet between more than one PC.
I connect to a SMC7004VWBR router/firewall wirelessly, but well enough for skype. I would be using my computer as a gateway, of sorts, with this device?
Not normally. That router is the gateway to the internet for everything on your LAN. It's a standalone device, and you don't need any computers running to use it.
That is much the best way to do VOIP. You don't need a power-wasting PC running all the time.
The device wouldn't work with Gizmo, then? I'd need a VoIP provider, along the lines of vonage?
As the other person said: You need a VOIP provider, just pick one that you're happy with. Start off by looking for one that publishes set up instructions where you can find them before having to sign up. There are some that are set up to work with some pre-configured devices, and don't provide information for using other devices.
Another gotcha is how do you call someone on another VOIP network? How do you call johndoe@sip.example.com from an ordinary telephone plugged into a VOIP box? Or even some softphones? Likewise for numerical user IDs, the same number can be used on different providers.
On Thu, 02 Nov 2006 09:22:46 +1030, Tim wrote: [...]
I connect to a SMC7004VWBR router/firewall wirelessly, but well enough for skype. I would be using my computer as a gateway, of sorts, with this device?
Not normally. That router is the gateway to the internet for everything on your LAN. It's a standalone device, and you don't need any computers running to use it.
That is much the best way to do VOIP. You don't need a power-wasting PC running all the time.
While it may be the best way to do VoIP, it's not an option for me, as I don't have physical access to the router :(
[...]
As the other person said: You need a VOIP provider, just pick one that you're happy with. Start off by looking for one that publishes set up instructions where you can find them before having to sign up. There are some that are set up to work with some pre-configured devices, and don't provide information for using other devices.
Another gotcha is how do you call someone on another VOIP network?
[...]
Whatever it is that a softphone, such as Skype, is doing, I want to do with hardware. I don't need the functionality described.
Is it possible to configure the computer and the ATA to make VoIP calls, such as I currently do with Skype? I've looked at the Skype phones which plug into the USB port of a computer, but don't like that option.
thanks,
Thufir
On Tue, 31 Oct 2006 19:43:34 -0500, Frank Pineau wrote: [...]
What does this have to do with Fedora?
I'm trying to determine what networking problems I might run into due to connecting the device to a computer and not a router.
In the user manual for the adapter (in the "Advanced Configuration" section), though, I note there is a configuration screen where you enter the SIP server, outbound proxy, and SIP user ID and password. Did you try reading the manual?
Ah, I didn't see that part. Pardon.
-Thufir
Thufir:
I connect to a SMC7004VWBR router/firewall wirelessly, but well enough for skype. I would be using my computer as a gateway, of sorts, with this device?
Tim:
Not normally. That router is the gateway to the internet for everything on your LAN. It's a standalone device, and you don't need any computers running to use it.
Thufir:
While it may be the best way to do VoIP, it's not an option for me, as I don't have physical access to the router :(
In which case, you'd need two network cards in your PC (one of which could be your wireless). And you'd share out to the second one, where you connect the VOIP box.
Though, depending on the system, that still mightn't work. Your router mightn't pass through network connections in the manner required for VOIP to work.
Another gotcha is how do you call someone on another VOIP network?
Whatever it is that a softphone, such as Skype, is doing, I want to do with hardware. I don't need the functionality described.
Ah, but how would you call someone who wasn't using Skype? That's where this VOIP thing falls apart. If you want to call someone who uses Skype, you need it as well, and vice versa. Then how many other systems do you also have, for your friends who use something that's not Skype? It's the same madness as ICQ vs MSN vs Yahoo vs Jabber... They don't talk to each other, and you'll never convince some people to use a different system. It's a bit shortsighted to think that you won't need anything else than Skype.
Is it possible to configure the computer and the ATA to make VoIP calls, such as I currently do with Skype?
If you're going to use the computer, you have a couple of options, at least:
Buy the ATA which is a network device between *something* and an ordinary telephone. Configure your networking for it to work.
Use a softphone and your soundcard.
On Tue, 31 Oct 2006 19:43:34 -0500, Frank Pineau wrote: [...]
What does this have to do with Fedora?
In the user manual for the adapter (in the "Advanced Configuration" section), though, I note there is a configuration screen where you enter the SIP server, outbound proxy, and SIP user ID and password. Did you try reading the manual?
'5. How do I setup my Grandstream Phone for go2call network? typical configuration is: SIP Server: voip01.go2call.com Outbound proxy: (Should leave it blank, because it's a GW) User ID: xxxxx (your Go2Call PIN number) Authentication ID: same as your User ID Password: xxxxxxx (Your Go2Call password) NAT Traversal: YES (WITHOUT setting the STUN server) 6. How do I setup my Grandstream Phone for FWD service? typical configuration is: SIP Server: fwd.pulver.com outbound proxy: 192.246.69.247:5082 (used only when behind firewall, otherwise leave it blank) User ID: xxxxxx (your FWD account number) Authentication/Login ID: xxxxx (same as above, your FWD account number) Password: xxxxx (your FWD password) NAT Traversal: No (You need to set up your STUN server if you don't have outbound proxy)'
http://www.grandstream.com/FAQ.pdf
I was wondering how the ATA knew what to do. I didn't know that there was a setting, such as "SIP server". I was wondering how the device knew what do. Knowing what to look for, I now know the generalities :)
thanks,
Thufir
On 11/1/06, Tim ignored_mailbox@yahoo.com.au wrote:
Ah, but how would you call someone who wasn't using Skype? That's where this VOIP thing falls apart. If you want to call someone who uses Skype, you need it as well, and vice versa. Then how many other systems do you also have, for your friends who use something that's not Skype? It's the same madness as ICQ vs MSN vs Yahoo vs Jabber... They don't talk to each other, and you'll never convince some people to use a different system. It's a bit shortsighted to think that you won't need anything else than Skype.
That is the beauty behind SIP providers like Gizmo and FWD. They openly peer with one another so you can call other users from a different provider just by using an "area code".
That is the thing that sucks about Vonage (eventhough it uses SIP too) and Skype ... they don't provide these interconnections.
On Thursday 02 November 2006 01:33, Michael Wiktowy wrote:
On 11/1/06, Tim ignored_mailbox@yahoo.com.au wrote:
Ah, but how would you call someone who wasn't using Skype? That's where this VOIP thing falls apart. If you want to call someone who uses Skype, you need it as well, and vice versa. Then how many other systems do you also have, for your friends who use something that's not Skype? It's the same madness as ICQ vs MSN vs Yahoo vs Jabber... They don't talk to each other, and you'll never convince some people to use a different system. It's a bit shortsighted to think that you won't need anything else than Skype.
That is the beauty behind SIP providers like Gizmo and FWD. They openly peer with one another so you can call other users from a different provider just by using an "area code".
That is also the achilles heel of SIP. I've now signed up for several of these providers, but the confirming email containing the connection details never arrives, and there is no place where one can actually contact a human to straighten things out, AND their software won't allow you to repeat the process. My guess is that verizon is (its a fact they filter according to their broken definition of spam and you can't opt out without changing your email address) out these emails "from the competition" to preserve their stranglehold on their long distance income. Because of this, I have yet to make a phone call with anything but skype.
If someone knows howto make ekiga work, without having to screw around with the confirming emails, please educate me.
That is the thing that sucks about Vonage (eventhough it uses SIP too) and Skype ... they don't provide these interconnections.
And that will be their achilles heel. All the tv adds notwithstanding, until I can call anybody, and there is a directory service that works across all these providers, its all going to be an also ran at the end of the month.
On Thu, 02 Nov 2006 13:19:44 +1030, Tim wrote: [...]
Thufir:
While it may be the best way to do VoIP, it's not an option for me, as I don't have physical access to the router :(
In which case, you'd need two network cards in your PC (one of which could be your wireless). And you'd share out to the second one, where you connect the VOIP box.
This is the setup I have ready to go, I'm just holding off on buying the VoIP hardware until I have a better handle on any potential problems.
Though, depending on the system, that still mightn't work. Your router mightn't pass through network connections in the manner required for VOIP to work.
That'd be a NAT problem? Or something else?
Right, I'm narrowing down potential problems. I think the biggest, for me, might be:
'What is STUN?
STUN stands for Simple Traversal of UDP (User Datagram Protocol) over NAT. It is a protocol which enables your IP phone to detect the presence and type of NAT behind which the phone is placed. An IP phone that supports STUN can intelligently modify the private IP address and port in its SIP/SDP message by using the NAT mapped public IP address and port through a series of STUN queries against a STUN server located on the public Internet. This will allow SIP signaling and RTP media to successfully traverse a NAT without requiring any configuration changes on the NAT.
Was this answer helpful? Yes No ' http://www.freeworlddialup.com/help/?p=knowledgebase&c=10&a=47
I think that my situation may require STUN. I don't require asterix? Asterix wouldn't help? There's some mention of asterix bypassing NAT issues with another standard (can't find the link at the moment).
Because I'm behind a firewall and am not the sysadmin, I have to be prepared for port forwarding problems, or other, normally easily fixed problems.
-Thufir
On Thu, Nov 02, 2006 at 02:08:35 -0500, Gene Heskett gene.heskett@verizon.net wrote:
And that will be their achilles heel. All the tv adds notwithstanding, until I can call anybody, and there is a directory service that works across all these providers, its all going to be an also ran at the end of the month.
There are already distributed directories being built by people. For a lot of uses you don't need a directory anyway, the person you want to talk to will have given you your contact information.
There are already open systems that people can run on an internet connection that will allow anyone in a similar situation to peer with them. Asterisk is probably the most well known system in this area. That doesn't get you a POTS connection. For that you need a connection from your local phoone company or access to a gateway over the internet. Eventually everyone will be connected peer to peer, but not any time soon.
On Thu, 2006-11-02 at 02:08 -0500, Gene Heskett wrote:
the achilles heel of SIP. I've now signed up for several of these providers, but the confirming email containing the connection details never arrives
Have you tried signing up using a webmail service provider? Your mail won't pass through the hands of your ISP, that way.
Off the top of my head, I know of Fastmail.fm, mail.yahoo.com, or even the dreaded Hotmail... Alternatively, if you have your own domain name, like I do, you might be able to use them for your SMTP and POP services.
Tim:
Though, depending on the system, that still mightn't work. Your router mightn't pass through network connections in the manner required for VOIP to work.
Thufir:
That'd be a NAT problem? Or something else?
Yes, and possible firewalling, as well (at the router, as well as the PC).
Right, I'm narrowing down potential problems. I think the biggest, for me, might be:
'What is STUN?
STUN stands for Simple Traversal of UDP (User Datagram Protocol) over NAT. It is a protocol which enables your IP phone to detect the presence and type of NAT behind which the phone is placed. An IP phone that supports STUN can intelligently modify the private IP address and port in its SIP/SDP message by using the NAT mapped public IP address and port through a series of STUN queries against a STUN server located on the public Internet. This will allow SIP signaling and RTP media to successfully traverse a NAT without requiring any configuration changes on the NAT.
Or, otherwise, very simply described as a TUNNEL through the barriers in your network.
Quite a few VOIP providers run their own STUN service, since so many people are behind NAT that they'd *have* to. Every one that I've played with, at least, has.
It's much like using a proxy with your web browser, it's just another address you configure into the VOIP setup. There's a service in the middle connecting you together. Though, in this case, it's just used to initiate the call. After that, you're directly connected to each other.
STUN will get you through NAT, in most cases. A firewall, as well, might be a problem. You'd have to try it and see.
Being behind NAT and/or a firewall does mean you'll need to use a VOIP provider. If you had none of them in your way, or complete control over them, you could run as your own service (the same as how people run their own web and mail services, for instance).
On Thu, 2006-11-02 at 01:08, Gene Heskett wrote:
That is the thing that sucks about Vonage (eventhough it uses SIP too) and Skype ... they don't provide these interconnections.
And that will be their achilles heel. All the tv adds notwithstanding, until I can call anybody, and there is a directory service that works across all these providers, its all going to be an also ran at the end of the month.
I've heard good things about http://www.sunrocket.com/. I don't have any personal experience or connection with them.
On 11/2/06, Gene Heskett gene.heskett@verizon.net wrote:
That is also the achilles heel of SIP. I've now signed up for several of these providers, but the confirming email containing the connection details never arrives, and there is no place where one can actually contact a human to straighten things out, AND their software won't allow you to repeat the process. My guess is that verizon is (its a fact they filter according to their broken definition of spam and you can't opt out without changing your email address) out these emails "from the competition" to preserve their stranglehold on their long distance income. Because of this, I have yet to make a phone call with anything but skype.
If someone knows howto make ekiga work, without having to screw around with the confirming emails, please educate me.
It sounds like you have more of a problem with your ISP than anything else. Most registrations for service require some sort of email verification even if they are free. It is a necessary evil. I would suggest getting a webmail address if Verizon is pulling shenanigans like that. Unfortunately, if your provider is blocking registration emails, you are presented with a catch-22. One method of getting around that is to use a throw-away email account generator like http://www.mailinator.com/ to either get a more permanent webmail address or for the service registration itself.
Ekiga is quite good in that it allows me to simultaneously login to multiple SIP accounts at once and I have used it fine with Gizmo and FWD as well as the SIP account that you can get through ekiga.net . There is some good info on setting up ekiga with ekiga.net or some other SIP provider in the manual at http://ekiga.org/documentation/ekiga.pdf
/Mike
On Fri, 03 Nov 2006 02:08:37 +1030, Tim wrote: [...]
Being behind NAT and/or a firewall does mean you'll need to use a VOIP provider. If you had none of them in your way, or complete control over them, you could run as your own service (the same as how people run their own web and mail services, for instance).
[...]
This is where http://en.wikipedia.org/wiki/IAX would come into play, as a workaround?
-Thufir
Tim:
Being behind NAT and/or a firewall does mean you'll need to use a VOIP provider. If you had none of them in your way, or complete control over them, you could run as your own service (the same as how people run their own web and mail services, for instance).
Thufir:
This is where http://en.wikipedia.org/wiki/IAX would come into play, as a workaround?
Yes, and no. Yes, you could do something like that when your network is entirely yours to control. No, I'd probably call it a solution, rather than a workaround. ;-) I'd call *having* to be a peer on someone else's VOIP network a workaround to networking problems.
On Fri, 03 Nov 2006 12:00:45 +1030, Tim wrote: [...]
Thufir:
This is where http://en.wikipedia.org/wiki/IAX would come into play, as a workaround?
Yes, and no. Yes, you could do something like that when your network is entirely yours to control. No, I'd probably call it a solution, rather than a workaround. ;-) I'd call *having* to be a peer on someone else's VOIP network a workaround to networking problems.
[...]
I would, or might, need admin access to the router/firewall?
-Thufir
On Fri, 2006-11-03 at 03:29 +0000, Thufir wrote:
I would, or might, need admin access to the router/firewall?
If you want to run any server that accepts connections through your firewall, you need to open ports through it, and possibly forward ports through. It depends on the system, some modem/routers don't act as a firewall, some aren't enabled, some will try to automatically route connections through (I've seen one that uses NAT, and does that for you).
On 11/3/06, Tim ignored_mailbox@yahoo.com.au wrote:
On Fri, 2006-11-03 at 03:29 +0000, Thufir wrote:
I would, or might, need admin access to the router/firewall?
If you want to run any server that accepts connections through your firewall, you need to open ports through it, and possibly forward ports through. It depends on the system, some modem/routers don't act as a firewall, some aren't enabled, some will try to automatically route connections through (I've seen one that uses NAT, and does that for you).
That is the whole reason for STUN AFAIK. It uses a STUN server on the outside of the firewall to negotiate a path back to two peers that are behind a NAT firewall using the inherent NAT inner workings so that the peers can talk to each other directly. http://en.wikipedia.org/wiki/STUN All SIP providers and clients that I have experience with use this method for accepting calls through a NAT withtou having to administer the NAT firewall. GTalk (jingle/jabber based) also uses this method.
/Mike
On Fri, 03 Nov 2006 11:20:25 -0500, Michael Wiktowy wrote: [...]
That is the whole reason for STUN AFAIK. It uses a STUN server on the outside of the firewall to negotiate a path back to two peers that are behind a NAT firewall using the inherent NAT inner workings so that the peers can talk to each other directly. http://en.wikipedia.org/wiki/STUN All SIP providers and clients that I have experience with use this method for accepting calls through a NAT withtou having to administer the NAT firewall. GTalk (jingle/jabber based) also uses this method.
/Mike
Thanks for making the wikipedia entry easier to digest :)
I don't mind buying the hardware and monkeying around with it to get things to work, but I don't want to waste my time. Can I somehow simulate STUN? Some sort of advanced ping?
thanks,
Thufir
On 11/3/06, Thufir hawat.thufir@gmail.com wrote:
I don't mind buying the hardware and monkeying around with it to get things to work, but I don't want to waste my time. Can I somehow simulate STUN? Some sort of advanced ping?
The SIP hardware that I have seen has STUN built into it. No doubt not all of them do.
However, according to the Gizmo forums, that particular SIP hardware adapter that you are looking into has a problem with the STUN feature that oddly enough requires you to clear it to work ... FYI: http://forum.gizmoproject.com/viewtopic.php?t=2295
Here are the details that you have to manually cram into the settings in the first place: http://support.gizmoproject.com/index.php?_a=knowledgebase&_j=questionde...
Tim:
If you want to run any server that accepts connections through your firewall, you need to open ports through it, and possibly forward ports through. It depends on the system, some modem/routers don't act as a firewall, some aren't enabled, some will try to automatically route connections through (I've seen one that uses NAT, and does that for you).
Michael Wiktowy:
That is the whole reason for STUN AFAIK. It uses a STUN server on the outside of the firewall to negotiate a path back to two peers that are behind a NAT firewall using the inherent NAT inner workings so that the peers can talk to each other directly.
Yes, that's what you do with "clients", I was talking about running "servers".
http://en.wikipedia.org/wiki/STUN All SIP providers and clients that I have experience with use this method for accepting calls through a NAT withtou having to administer the NAT firewall. GTalk (jingle/jabber based) also uses this method.
As I'd said previously.
On Fri, 03 Nov 2006 15:31:32 -0500, Michael Wiktowy wrote: [...]
However, according to the Gizmo forums, that particular SIP hardware adapter that you are looking into has a problem with the STUN feature that oddly enough requires you to clear it to work ... FYI: http://forum.gizmoproject.com/viewtopic.php?t=2295
Here are the details that you have to manually cram into the settings in the first place: http://support.gizmoproject.com/index.php?_a=knowledgebase&_j=questionde...
Thanks for the link. It's a rainy day, and I'm thinking about going down to best buy to buy a skype phone and booting windows, to test out the general idea. With skype, there's, amazingly, a USB phone which supposedly works with linux : http://support.a-link.com/phonemate/IPU1.htm.
The ones at Best Buy require windows, but I might try it out just to see how things go. Either way I'll probably return the skype phone. If the skype phone works I'll keep looking into a SIP hardware phone or ATA.
thanks,
Thufir
-
Bruno Wolff III -
Frank Pineau -
Gene Heskett -
Les Mikesell -
Michael Wiktowy -
Thufir -
Tim