Monday, July 11, 2005
VoIP USA
I took the covered wagon from Minneapolis up to Fargo, North Dakota (it was a DC9 actually, a bit late and very hot as it was 90 °F and sunny).
After visiting Walmart - 8pm on a Sunday - BestBuy, OfficeMax, OfficeDepot and finally Radio Shack (Tandy as was in the UK) I accumulated enough bits of wire to power and connect my BT Broadband Voice box (Cisco ATA186). I am connecting the laptop wirelessly to the hotel hi-speed internet and a crossover ethernet cable connects to the ATA186. The laptop is on an external IP address and the ATA gets a 192.168.0.x address from the laptop. I took a photo but seem to need yet another bit of wire to get it onto the laptop - there is no GPRS signal for the Treo600 to email me the photo it took. Watch this space. Picture here shows laptop on left connected by crossover cable to ATA in centre with hotel phone on right. Can't get the in-line image upload to work :(
The system works a treat, I accidentally called someone in the UK at 03:30 (woops) but then saved an arm and a leg by collecting my mobile voicemail at BT's offpeak Orange rate of 11.4p/minute rather than making a roaming Orange call at £1/min or whatever.
This was my first "hotspot" experiment - how to hook a dumb ATA up to a service that requires authentication via a web page. The laptop handled the authentication and hid the ATA from the hotspot - but how would you do this with an 802.11b SIP phone ??
I phoned across town via the UK to my friend in Fargo who has cable internet and a Vonage VoIP service (2 lines) with no analogue phoneline. There was a marginal delay but not the sort that makes conversation difficult. His ATA is the Linksys PAP2 but locked down to the Vonage service.
In Best Buy there were several routers or ATAs on offer with bundled VoIP service. Generally you get a mail-in rebate or account credit equal to the hardware cost when you sign up for the service. The kit included ATAs and routers with VoIP phone ports in both wired and wireless flavours.
Clearly the market focus is on getting cable internet users (the vast majority) to use VoIP in preference to analogue PSTN phone lines. The monthly cost is much lower and a bundle of services like voicemail are included for free.
After visiting Walmart - 8pm on a Sunday - BestBuy, OfficeMax, OfficeDepot and finally Radio Shack (Tandy as was in the UK) I accumulated enough bits of wire to power and connect my BT Broadband Voice box (Cisco ATA186). I am connecting the laptop wirelessly to the hotel hi-speed internet and a crossover ethernet cable connects to the ATA186. The laptop is on an external IP address and the ATA gets a 192.168.0.x address from the laptop. I took a photo but seem to need yet another bit of wire to get it onto the laptop - there is no GPRS signal for the Treo600 to email me the photo it took. Watch this space. Picture here shows laptop on left connected by crossover cable to ATA in centre with hotel phone on right. Can't get the in-line image upload to work :(
The system works a treat, I accidentally called someone in the UK at 03:30 (woops) but then saved an arm and a leg by collecting my mobile voicemail at BT's offpeak Orange rate of 11.4p/minute rather than making a roaming Orange call at £1/min or whatever.
This was my first "hotspot" experiment - how to hook a dumb ATA up to a service that requires authentication via a web page. The laptop handled the authentication and hid the ATA from the hotspot - but how would you do this with an 802.11b SIP phone ??
I phoned across town via the UK to my friend in Fargo who has cable internet and a Vonage VoIP service (2 lines) with no analogue phoneline. There was a marginal delay but not the sort that makes conversation difficult. His ATA is the Linksys PAP2 but locked down to the Vonage service.
In Best Buy there were several routers or ATAs on offer with bundled VoIP service. Generally you get a mail-in rebate or account credit equal to the hardware cost when you sign up for the service. The kit included ATAs and routers with VoIP phone ports in both wired and wireless flavours.
Clearly the market focus is on getting cable internet users (the vast majority) to use VoIP in preference to analogue PSTN phone lines. The monthly cost is much lower and a bundle of services like voicemail are included for free.
Sunday, July 10, 2005
From South Witham to Minneapolis airport
This comes to you via the wireless internet access at MSP airport, a pay-to-use service which I access via IPASS roaming thanks to a prepay facility run by the folks at http://www.roamintl.com/
Yesterday's VoIP-a-thon at S Witham was interesting. A collection of many different ATAs, IP phones and Skype over bluetooth headsets. Basically it all "just worked" via Tom's wireless network, which runs on beer and tobacco. http://www.wireless.southwitham.net/
My PAP2 worked good on two accounts, also the Cisco BTBBV box fired up nicely on the SWBB net once their MAC codes were granted access by Tom. NO port forwarding or firewall fiddling required. We also dipped our toes in the confusopoly of charging rates to different types of number / different countries.
Best go now - off to queue behind a "line" of Americans taking their shoes off to go through a metal detector (?).
Yesterday's VoIP-a-thon at S Witham was interesting. A collection of many different ATAs, IP phones and Skype over bluetooth headsets. Basically it all "just worked" via Tom's wireless network, which runs on beer and tobacco. http://www.wireless.southwitham.net/
My PAP2 worked good on two accounts, also the Cisco BTBBV box fired up nicely on the SWBB net once their MAC codes were granted access by Tom. NO port forwarding or firewall fiddling required. We also dipped our toes in the confusopoly of charging rates to different types of number / different countries.
Best go now - off to queue behind a "line" of Americans taking their shoes off to go through a metal detector (?).
Monday, July 04, 2005
Linksys PAP2 - out and about.
I thought I would try to see how little configuration I could get away with. A couple of posts by others suggested port forwarding to the PAP2 wasn't necessary and indeed after turning it off in my router (Belkin wireless ADSL jobbie) I can confirm this. Equally I don't have any ports for RTP streams open, and no STUN server settings.
I rebooted the beast this morning and it still worked, then rebooted the router too and that didn't affect it. Then I took it to the office and hooked it up behind an Intertex IX66 router without doing any port openings or forwarding - and it worked there too ! Granted some ports may have been open from previous VoIP efforts. So pretty robust and portable really.
All in all a success. The office DECT phone (BT Diverse 5310 - Siemens) would ring on incoming calls without the microfilter, it also showed the caller identity so CLID works fine too both in and out.
One thing that can catch you out is the lack of refresh on the Info page of the web interface - clicking on Info doesn't refresh it, it needs a browser refresh F5 to bring it up to date and show the current time and call / line status.
Its a bit clunky having a local 01780 incoming number as you can dial the VoIP number from BT locally without the 01780 code but to dial out over VoIP always needs the full dial code and number. I think a tweaking of the Dial Plan could fix this - insert 01780 in front of any number dialled that starts with a 7. Something else to learn how to do.
For VoIP to work in consumer land I think the ATA needs to be pre-configured to the account and tested before shipping. NAT routers could cause enough headaches without the poor user having to learn to setup the ATA.
I can see why people are shipping ADSL modem/routers with VoIP functions built in - better Quality of Service and no NAT workaround issues. Sipgate are selling a router / ATA from AVM for £120 that provides two analogue phone VoIP ports as well as the usual ADSL modem/router functions. Appears to have some integration of the analogue PSTN phone line too.
I rebooted the beast this morning and it still worked, then rebooted the router too and that didn't affect it. Then I took it to the office and hooked it up behind an Intertex IX66 router without doing any port openings or forwarding - and it worked there too ! Granted some ports may have been open from previous VoIP efforts. So pretty robust and portable really.
All in all a success. The office DECT phone (BT Diverse 5310 - Siemens) would ring on incoming calls without the microfilter, it also showed the caller identity so CLID works fine too both in and out.
One thing that can catch you out is the lack of refresh on the Info page of the web interface - clicking on Info doesn't refresh it, it needs a browser refresh F5 to bring it up to date and show the current time and call / line status.
Its a bit clunky having a local 01780 incoming number as you can dial the VoIP number from BT locally without the 01780 code but to dial out over VoIP always needs the full dial code and number. I think a tweaking of the Dial Plan could fix this - insert 01780 in front of any number dialled that starts with a 7. Something else to learn how to do.
For VoIP to work in consumer land I think the ATA needs to be pre-configured to the account and tested before shipping. NAT routers could cause enough headaches without the poor user having to learn to setup the ATA.
I can see why people are shipping ADSL modem/routers with VoIP functions built in - better Quality of Service and no NAT workaround issues. Sipgate are selling a router / ATA from AVM for £120 that provides two analogue phone VoIP ports as well as the usual ADSL modem/router functions. Appears to have some integration of the analogue PSTN phone line too.
Sunday, July 03, 2005
Linksys PAP2-NA - Day 2
After inspecting the firewall logs I could see incoming traffic bouncing off the firewall that was from sipgate IP addresses and on SIP ports, so I put the Linksys into the DMZ on the router exposing it to the ills of the internet.
Calling it from the landline I got ringing tone but the DECT phone on the Linksys didn't ring, and there was no flashing LEDs etc to indicate activity. On inspecting the Info page I could however see that it was ringing. Inserting an ADSL microfilter in between the DECT phone and the PAP2 brought it to life - clearly the output needs an RJ11-BT adaptor that includes a ringing capacitor (provided in my case by the microfilter). So it is actually working with Sipgate on Line 1 while in the DMZ !!
Without the ring capacitor you can answer the call if you know its ringing, but that is only evident from the web interface :-(
Reading up about the Linksys PAP2-NA I discover that it is allegedly the same unit as a SIPURA SPA2000
The forums at http://voxilla.com/PNphpBB2.html were very helpful, and include configuration tools for common ATAs and poplar VoIP service providers, very useful !
I turned off the DMZ on the router and it still works. Not sure it will survive its hourly re-registration though. I have port forwarding of 5060 and 5061 to the Linksys but no other router tweaks. May need to test its resilience.
Inspired by getting the first line to work I set about connecting the second up to beta VoIP service from 18866.co.uk - this sat saying "can't connect to registration server" and declined to do anything - no blue LED :-(
After messing around with Line 1 for a while I looked at the settings in Line 2 and saw that they were corrupted - what should have been "sip.call18866.co.uk" in the Proxy field had morphed into something like "985@-16,1428@-16,1777" which is a tone definition or dial plan config - so the moral here is to check what is actually saved when you hit "Save" by revisiting the same page.
Corrected the settings and Line 2 leapt into life, made a call out and all worked well. I tested it by ringing my answering machine, to ensure 2-way audio was present.
Cool - 4 blue LEDs all shining nicely at me. Lets see if it lasts. Still not in the DMZ.
I found a load of NAT and STUN settings at the bottom of the SIP page when viewed in Advanced mode as Admin. Currently they are blank however the voxilla config wizard gives the settings for sipgate. I'm never quite happy with these 2-line devices having one set of parameters for something, nagging doubts that it'll work for both providers. For reference I have Firmware Version: 2.0.12(LS)
Off to buy some cake to celebrate.
Calling it from the landline I got ringing tone but the DECT phone on the Linksys didn't ring, and there was no flashing LEDs etc to indicate activity. On inspecting the Info page I could however see that it was ringing. Inserting an ADSL microfilter in between the DECT phone and the PAP2 brought it to life - clearly the output needs an RJ11-BT adaptor that includes a ringing capacitor (provided in my case by the microfilter). So it is actually working with Sipgate on Line 1 while in the DMZ !!
Without the ring capacitor you can answer the call if you know its ringing, but that is only evident from the web interface :-(
Reading up about the Linksys PAP2-NA I discover that it is allegedly the same unit as a SIPURA SPA2000
The forums at http://voxilla.com/PNphpBB2.html were very helpful, and include configuration tools for common ATAs and poplar VoIP service providers, very useful !
I turned off the DMZ on the router and it still works. Not sure it will survive its hourly re-registration though. I have port forwarding of 5060 and 5061 to the Linksys but no other router tweaks. May need to test its resilience.
Inspired by getting the first line to work I set about connecting the second up to beta VoIP service from 18866.co.uk - this sat saying "can't connect to registration server" and declined to do anything - no blue LED :-(
After messing around with Line 1 for a while I looked at the settings in Line 2 and saw that they were corrupted - what should have been "sip.call18866.co.uk" in the Proxy field had morphed into something like "985@-16,1428@-16,1777" which is a tone definition or dial plan config - so the moral here is to check what is actually saved when you hit "Save" by revisiting the same page.
Corrected the settings and Line 2 leapt into life, made a call out and all worked well. I tested it by ringing my answering machine, to ensure 2-way audio was present.
Cool - 4 blue LEDs all shining nicely at me. Lets see if it lasts. Still not in the DMZ.
I found a load of NAT and STUN settings at the bottom of the SIP page when viewed in Advanced mode as Admin. Currently they are blank however the voxilla config wizard gives the settings for sipgate. I'm never quite happy with these 2-line devices having one set of parameters for something, nagging doubts that it'll work for both providers. For reference I have Firmware Version: 2.0.12(LS)
Off to buy some cake to celebrate.
Saturday, July 02, 2005
Linksys PAP-2 experiences
An "as it happened" walkthrough of setting up a Linkysys PAP-2 :-
The Linksys PAP2 is an analogue telephone adapter (ATA) to allow a normal phone to make Voice over IP (VoIP) calls. It has an ethernet socket to connect to a broadband router and two RJ11 phone sockets. Wall mounting screw holes are provided, as is a plastic foot to allow the unit to sit vertically on a surface. At 4 inches (100mm) square by an inch (30mm) thick it isn't large.
I bought mine from http://www.broadbandbuyer.co.uk/ for just over £40 inc VAT and p&p. The price was the main reason for selecting it over the Sipura models.
The box says model PAP2-UK but the device PAP2-NA, as the quick installation sheet says to dial "1 + area code + number" it would appear to be a USA model localised for the UK by adding a clip-on 13A plug for the power supply and a few sheets of CE compliance information.
On plugging in to the LAN and powering up I had one intense blue LED pulsing away with the letter D in morse ie -.. -.. this is the power LED. Not quite the "all LEDs will be solidly lit" of the documentation. I had to find an RJ11 to BT adapter so I could plug in the DECT phone I planned to use.
After a bit of thought I realised that the LAN LED was not lit and nor was the other end on the router, so tried a different cable and got the LAN LED up and the power LED stopped flashing - one step forward ! Reverted to original cable after a bit of wiggling plug in socket.
I see from the router that I can ping the PAP2 on the IP address it collected by DHCP. Popping the relevant IP address into Firefox I get to the Linksys PAP2 configuration page. Under Line 1 it correctly shows the Hook State: as "On" and it toggles in response to me keying the DECT phone. So far so good. There is however no dial tone, so I will RTFM and set it up as guided.......
First tried the phone keypad tone IVR menu - punch **** into the phone keypad and it responds verbally, hang up to exit. So I know the phone connection works, and where to go if I needed to set up the IP address manually. 110# discloses the current IP address, 100# says if DHCP is enabled or not. 732668# (reboot#) and 73738# (reset# - a factory reset) might be handy later.
Back to the web interface - this has Basic View and Advanced View modes, and runs in Admin and User modes (the upper right of the screen says User Login when in Admin mode, and vice versa). Default is to have no passwords, though the CD manual refers to setting passwords via the IVR system. This looks a bit clunky - "To enter A, B, C, a, b, or c — press 2" - however its actually just using a PIN as to put in Phil you would enter 7445, its not like texting and pressing the 2 key 5 times to get a 'b'.
Clearly the assumption in the manual is that the unit is shipped pre-configured with a voice service, as the last chapter of the manual concludes
"Configuring the Settings for Your Internet Phone Service
If you want to change the settings for your Internet phone service, visit the website of your Internet phone service provider and make configuration changes online. For more information, refer to the instructions provided by your Internet phone service provider."
The IVR menu command 7932# is to enable/disable the web interface, this is where a password would be required if a service provider had shipped the unit locked down to their service.
So, another cup of coffee then let's try to figure out the setup :-)
In the User mode, Basic or Advanced view, there are 4 screens on the web interface titled Info System User1 User2. Info is a Status screen, System covers TCP/IP network settings and password and the User screens are full of Speed dial, Suplementary services, Distinctive ring and othe ruser features.
Change to Admin mode and extra screens become available - SIP, Regional, Line 1 and Line 2. Sounds like I'm going the right way. In the Advanced view version a further screen - Provisioning - appears. Checking back at some of the other screens I can see that Advanced really is when it comes to things like defining dial tones and codec pre-selection strings. Let's stick to basic for now....
Under Regional I select GMT +1 for the timezone and put in the date in mm/dd/yyyy format and the time in hh/mm. The screen actually specifices mm/dd but that left me in 2003 ! There's a drop down box for setting FXS impedance - this is set at 600 (ohms ?) but the dropdown box has a myriad choice of ohms + capacitance values. If I find out what the UK phones expect I can change it later.
Line 1 looks to be the place for setting up the account details. I'll start with a Sipgate account from www.sipgate.co.uk as this gives me a free regional incoming number that matches my local dial code 01780. The PAP-2 isn't listed on Sipgate's help pages, so I look at the "other devices" page, then revert to the X-ten page where you can get a personalised setup screen pre-filled with your details (bit of a crutch for a newbie). Taking the minimalist approach I entered :-
Proxy: sipgate.co.uk
Display name : Phil Thompson
User ID : 140xxxx
Password : *****
the default SIP port (5060) and registration time (3600s) seemed to match sipgate's needs, so I hit "Save settings", get a "please wait page - bit of red and blue flickering from the power LED and, lo, the Line 1 blue LED is lit :-) and the Info page shows Registration: Online under Line 1 Status. Flip the DECT phone open, punch the green button and YES ! we have a dial tone.
The sipgate account page shows me online, so I dial up from the BT landline and the X-ten softphone on my PC leaps into action. Oh pooh. Two things registered on the same account.
Shut down X-lite, redial, "the service cannot be connected" message from BT. Let's try outbound - dial the sipgate 10000 test number and yes, a German guy talks back at me. So I have an outbound phone service but not incoming - sounds like a job for "Router Man".
The only port forwarding to the desktop is port 4569 which is IAX2 protocol, but I do know the X-lite was setup to use STUN in some way.
After hacking around I can get silence from BT when I try to call myself, but nothing else happens. Try again in the morning.
The Linksys PAP2 is an analogue telephone adapter (ATA) to allow a normal phone to make Voice over IP (VoIP) calls. It has an ethernet socket to connect to a broadband router and two RJ11 phone sockets. Wall mounting screw holes are provided, as is a plastic foot to allow the unit to sit vertically on a surface. At 4 inches (100mm) square by an inch (30mm) thick it isn't large.
I bought mine from http://www.broadbandbuyer.co.uk/ for just over £40 inc VAT and p&p. The price was the main reason for selecting it over the Sipura models.
The box says model PAP2-UK but the device PAP2-NA, as the quick installation sheet says to dial "1 + area code + number" it would appear to be a USA model localised for the UK by adding a clip-on 13A plug for the power supply and a few sheets of CE compliance information.
On plugging in to the LAN and powering up I had one intense blue LED pulsing away with the letter D in morse ie -.. -.. this is the power LED. Not quite the "all LEDs will be solidly lit" of the documentation. I had to find an RJ11 to BT adapter so I could plug in the DECT phone I planned to use.
After a bit of thought I realised that the LAN LED was not lit and nor was the other end on the router, so tried a different cable and got the LAN LED up and the power LED stopped flashing - one step forward ! Reverted to original cable after a bit of wiggling plug in socket.
I see from the router that I can ping the PAP2 on the IP address it collected by DHCP. Popping the relevant IP address into Firefox I get to the Linksys PAP2 configuration page. Under Line 1 it correctly shows the Hook State: as "On" and it toggles in response to me keying the DECT phone. So far so good. There is however no dial tone, so I will RTFM and set it up as guided.......
First tried the phone keypad tone IVR menu - punch **** into the phone keypad and it responds verbally, hang up to exit. So I know the phone connection works, and where to go if I needed to set up the IP address manually. 110# discloses the current IP address, 100# says if DHCP is enabled or not. 732668# (reboot#) and 73738# (reset# - a factory reset) might be handy later.
Back to the web interface - this has Basic View and Advanced View modes, and runs in Admin and User modes (the upper right of the screen says User Login when in Admin mode, and vice versa). Default is to have no passwords, though the CD manual refers to setting passwords via the IVR system. This looks a bit clunky - "To enter A, B, C, a, b, or c — press 2" - however its actually just using a PIN as to put in Phil you would enter 7445, its not like texting and pressing the 2 key 5 times to get a 'b'.
Clearly the assumption in the manual is that the unit is shipped pre-configured with a voice service, as the last chapter of the manual concludes
"Configuring the Settings for Your Internet Phone Service
If you want to change the settings for your Internet phone service, visit the website of your Internet phone service provider and make configuration changes online. For more information, refer to the instructions provided by your Internet phone service provider."
The IVR menu command 7932# is to enable/disable the web interface, this is where a password would be required if a service provider had shipped the unit locked down to their service.
So, another cup of coffee then let's try to figure out the setup :-)
In the User mode, Basic or Advanced view, there are 4 screens on the web interface titled Info System User1 User2. Info is a Status screen, System covers TCP/IP network settings and password and the User screens are full of Speed dial, Suplementary services, Distinctive ring and othe ruser features.
Change to Admin mode and extra screens become available - SIP, Regional, Line 1 and Line 2. Sounds like I'm going the right way. In the Advanced view version a further screen - Provisioning - appears. Checking back at some of the other screens I can see that Advanced really is when it comes to things like defining dial tones and codec pre-selection strings. Let's stick to basic for now....
Under Regional I select GMT +1 for the timezone and put in the date in mm/dd/yyyy format and the time in hh/mm. The screen actually specifices mm/dd but that left me in 2003 ! There's a drop down box for setting FXS impedance - this is set at 600 (ohms ?) but the dropdown box has a myriad choice of ohms + capacitance values. If I find out what the UK phones expect I can change it later.
Line 1 looks to be the place for setting up the account details. I'll start with a Sipgate account from www.sipgate.co.uk as this gives me a free regional incoming number that matches my local dial code 01780. The PAP-2 isn't listed on Sipgate's help pages, so I look at the "other devices" page, then revert to the X-ten page where you can get a personalised setup screen pre-filled with your details (bit of a crutch for a newbie). Taking the minimalist approach I entered :-
Proxy: sipgate.co.uk
Display name : Phil Thompson
User ID : 140xxxx
Password : *****
the default SIP port (5060) and registration time (3600s) seemed to match sipgate's needs, so I hit "Save settings", get a "please wait page - bit of red and blue flickering from the power LED and, lo, the Line 1 blue LED is lit :-) and the Info page shows Registration: Online under Line 1 Status. Flip the DECT phone open, punch the green button and YES ! we have a dial tone.
The sipgate account page shows me online, so I dial up from the BT landline and the X-ten softphone on my PC leaps into action. Oh pooh. Two things registered on the same account.
Shut down X-lite, redial, "the service cannot be connected" message from BT. Let's try outbound - dial the sipgate 10000 test number and yes, a German guy talks back at me. So I have an outbound phone service but not incoming - sounds like a job for "Router Man".
The only port forwarding to the desktop is port 4569 which is IAX2 protocol, but I do know the X-lite was setup to use STUN in some way.
After hacking around I can get silence from BT when I try to call myself, but nothing else happens. Try again in the morning.
