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This is a discussion on VOIP novice wishes to test first stage within the uk.telecom.voip forums, part of the Newsgroup Forums category; I'm still on dial-up here, but, in anticipation of having broadband shortly when I move to France, I'...
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I'm still on dial-up here, but, in anticipation of having broadband shortly when I move to France, I've bought a couple of inexpensive ATAs. I've programmed one of them on my LAN - just the basic bit, not the SIPs - so that (theoretically, as it seems to me, a complete newbie) someone should be able to ring my bell if they knew my IP address using the direct IP call system. I appreciate we might not be able to speak (remember, I'm on dial-up!), but, hey, it's Saturday night and I want my bell to ring! And it'll cost nothing! IP: 062255140001 (with the full-stops taken out, as my device expects) I'll unplug it if it gets too noisy, and report back. Forgive me if I'm talking through my <suitable voip-buzz-word>. John -- John Williams, Wirral, Merseyside, UK - no attachments to these addresses! Non-RISC OS posters change user to johnrwilliams or put 'risc' in subject for reliable contact! Who is John Williams? [url]http://www.picindex.info/author/[/url] |
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On 31 Mar, [email]UCEbin@tiscali.co.uk[/email] wrote:
[color=blue] > I've programmed one of them on my LAN - just the basic bit, not the SIPs - > so that (theoretically, as it seems to me, a complete newbie) someone > should be able to ring my bell if they knew my IP address using the direct > IP call system.[/color] OK - 30 minutes and nothing has happened, so evidently I've misunderstood the manual or everyone's out clubbing. My ATA is from DigiDave's, made in China, and cost GBP17: GATE 101. The manual says it can be used without an SIP proxy using public IP addresses. I /have/ purchased the adapters BT socket - RJ11 including ring circuit, and have plugged in a working BT-compatible phone. The ATA is plugged into my LAN, and I have AFAIK set the correct settings for Static IP - my public IP as reported by the script at: [url]http://www.picindex.info/iphost.php[/url] (it'll report yours for you), the IP of my device, the subnet mask, and two DNS server IPs. Is it unreasonable to expect my phone to ring? Have I misunderstood something(s)? If my expectations are unrealistic, forgive me. We all have to start somewhere! John -- John Williams, Wirral, Merseyside, UK - no attachments to these addresses! Non-RISC OS posters change user to johnrwilliams or put 'risc' in subject for reliable contact! Who is John Williams? [url]http://www.picindex.info/author/[/url] |
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On 2007-03-31, John Williams (News) <UCEbin@tiscali.co.uk> wrote:
[color=blue] > I'm still on dial-up here, but, in anticipation of having broadband shortly > when I move to France, I've bought a couple of inexpensive ATAs. > > I've programmed one of them on my LAN - just the basic bit, not the SIPs - > so that (theoretically, as it seems to me, a complete newbie) someone > should be able to ring my bell if they knew my IP address using the direct > IP call system.[/color] You have the correct idea. That should be possible. [color=blue] > I appreciate we might not be able to speak (remember, I'm on dial-up!), > but, hey, it's Saturday night and I want my bell to ring! And it'll cost > nothing! > > IP: 062255140001 > > (with the full-stops taken out, as my device expects) > > I'll unplug it if it gets too noisy, and report back. > > Forgive me if I'm talking through my <suitable voip-buzz-word>.[/color] Two corrections. Firstly, your IP is 62.255.140.11 so that number above should be 062255140011. Secondly, that is the format in which you input the IP to *send* from *your* ATA. Other ATAs use '*' to replace '.' and '@' when using IP-to-IP dialing from a handset. My understanding of SIP addressing is that username@IP should ring your phone, so it is necessary for you to configure a username on the device and let us know what it is. I admit to not understanding the GATE 101's method of using only the IP. There is the question of whether 62.255.140.11 is the address of the ATA or the router. If it's the router's IP you will have to have the telephony service on port 5060. Brian. |
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In article <eumtvs$sl1$1$8302bc10@news.demon.co.uk>, Brian <bandj@o2.co.uk>
wrote: [color=blue][color=green] > > I'm still on dial-up here, but, in anticipation of having broadband > > shortly when I move to France, I've bought a couple of inexpensive ATAs. > > > > I've programmed one of them on my LAN - just the basic bit, not the > > SIPs - so that (theoretically, as it seems to me, a complete newbie) > > someone should be able to ring my bell if they knew my IP address using > > the direct IP call system.[/color][/color] [color=blue] > You have the correct idea. That should be possible.[/color] [color=blue][color=green] > > I appreciate we might not be able to speak (remember, I'm on dial-up!), > > but, hey, it's Saturday night and I want my bell to ring! And it'll > > cost nothing! > > > > IP: 062255140001 > > > > (with the full-stops taken out, as my device expects) > > > > I'll unplug it if it gets too noisy, and report back. > > > > Forgive me if I'm talking through my <suitable voip-buzz-word>.[/color][/color] [color=blue] > Two corrections. Firstly, your IP is 62.255.140.11 so that number above > should be 062255140011.[/color] No, when I posted it was .1 - but it changes when the connection expires and is renewed. [color=blue] > Secondly, that is the format in which you input the IP to *send* from > *your* ATA.[/color] Ah - interesting [color=blue] > Other ATAs use '*' to replace '.' and '@' when using IP-to-IP dialing > from a handset.[/color] [color=blue] > My understanding of SIP addressing is that username@IP should ring your > phone, so it is necessary for you to configure a username on the device > and let us know what it is. I admit to not understanding the GATE 101's > method of using only the IP.[/color] It's an additional feature. I'm not yet registered with a SIP gateway supplier - there's no point until I get broadband. But I just wanted to test the ringing bit to see if it worked. I suspect the username@IP bit is part of the SIPs configuration, ie a stage further on than I am. And I expect that's what's invoked at the likes of Sipgate to connect one's calls. [color=blue] > There is the question of whether 62.255.140.11 is the address of the ATA > or the router. If it's the router's IP you will have to have the > telephony service on port 5060.[/color] The IP address I gave was correct at the time according to the PHP function $_SERVER["REMOTE_ADDR"] on one of my sites, and according to my 'router', which is a box running IPCop, as allocated by NTL. I deduced that when the router gets a call on port 5060, the default, it identifies the ATA as the appropriate network device to handle the request. I have been getting nuisance calls recently, and thought I heard a phone ring at 3:30 am, but my conventional landline caller ID monitor showed nothing. Either I dreamed it, or someone helpful rang the ATA at that time. I'd love to know which. I composed a reply to your posting, for which thanks, but seem to have lost it in a small crashette this morning. Present IP for another 1 hour 40 minutes approx is 62.255.144.10, and if anyone with a GATE 101 wants to give me a bell, that's: *** to get the voice menu 47 to select the IP option 062255144010 my current IP address Thanks again for input! Best wishes, John -- John Williams, Wirral, Merseyside, UK - no attachments to these addresses! Non-RISC OS posters change user to johnrwilliams or put 'risc' in subject for reliable contact! Who is John Williams? [url]http://www.picindex.info/author/[/url] |
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