Siemens C450IP/C460IP

Last weekend I was in PC World where I came upon a Philips cordless VOIP phone, but it had a major fatal flaw - it was a USB device. What I want is cordless VOIP phone that has an ethernet port and uses a protocol supported by Asterisk. Today, I discovered such a device exists - The Siemens Gigaset C450 IP, which will be the C460IP in the UK.

I'm not sure what the differences are between the 450 and 460 that merit what appears to be a ~£30 price difference (you can probably guess which is the more expensive model). I'm betting the only differences are UK Caller ID compatibility (it's a "dual mode" phone that can connect to the landline) and a power adaptor and phone line with UK plugs. If those are the only differences, I'll have to see if I can source the C450.

FDX 840

I've got my Fujitsu FDX-840 ATA up and running. It's an ex BT Broadband Voice ATA. It uses MGCP as it's protocol, apperently it can be converted to use SIP, but it seems to work fine with with Asterisk via MGCP, so I'll keep the current firmware.

There's a small amount of echo, and it seems to occur via both the landline and via Tesco internet.

SMS Part 3

I got out my Magic Messenger - that had trouble recieving texts as well, but I thought it was the device that was at fault. I decided to try and hook it up straight into the ADSL filter (ie, without going through any extension lines), so far it seems to be working - in as much as it is recieving texts - there's no easy way to tell if that's after several failed attemtps or not.

I think what I need to do is find a long ethernet cable so I can temporarily move the asterisk box next to the adsl filter and plug it straight in.

SMS Part 2

I'm still having trouble with SMS reception. I've modified the source to get a better sense of what's going wrong and it appears that I'm getting stop bit errors (it also looks like there are stop bit errors on transmit as well). One interesting thing is that one message did get through, which means reception is at least possible, so I'm thinking that tweaking the X100P's quality is required (but then, I need to do that for voice as well).

call through

I've been experimenting with call through - that's where I use my mobile phone to dial in via one of my (voip) lines and then get to call out on another. It works, though I really need to play about with the X110P to get the best quality that I can get out of it.

Caller ID & SMS

I've got Caller ID working. It took a small amount of effort, as the voip-user docs don't work right with the x100p (they're for the TDM400), but the docs on the lusyn page do work right.

I've also got SMS sending working, but I'm having a problem recieving. I can get it to go to the correct context and start the SMS app, all I get is

-- SMS TX 93 00 6D
-- SMS RX 93 00 6D
-- SMS TX 94 00 6C
-- SMS TX 92 01 FF 6E

I'm not certain what to do, as googling has not provided an answer.

Sipgate.co.uk

I've got sipgate.co.uk running. The main problem was realizing that the SIP ID isn't the phone number and the password isn't the password I use to access the website. In order to get the sipgate SIP-ID and password, you need to log in to the web site and at the bottom of the my account page is the needed connection information.

Gizmo Project

Thanks to the instructions at voip-info.org I've successfully connected asterisk to Gizmo Project.

Tesco's VOIP & Asterisk

UPDATE (2007-01-06) - I think the problems with Asterisk's registration expiring with Tesco is due to a bug in Asterisk. Have a look at my new post on the issue. You can find a link to the bug report I filed, which includes a patch against Asterisk 1.4.

Tesco's internet phone uses Asterisk's own protocol (IAX2), so you would think it should be relatively simple to get my asterisk server talking to it, and it almost is. I've been able to get my asterisk server and Tesco talking, by using the following:

iax.conf

[general]
bandwidth=low
diallow=lpc10
jitterbuffer=no
forcejitterbuffer=no
tos=lowdelay
;autokill=yes

register => TESCONUMBER:password@gateway.tescointernetphone.com

[TESCONUMBER]
type=friend
context=tesco
auth==md5
username=TESCONUMBER
secret=password
host=gateway.tescointernetphone.com
qualify=no


and the following in extensions.conf

[macro-outtesco]
exten => s,1,SetCallerID("TESCONUMBER" <TESCONUMBER>)
exten => s,2,Dial(IAX2/TESCONUMBER:password@gateway.tescointernetphone.com/${ARG1},${ARG2},r)
exten => s,3,Congestion


[tesco]
exten => INTTESCONUMBER9,1,Dial(SIP/scot)
exten => INTTESCONUMBER9,1,Answer()
exten => INTTESCONUMBER9,2,Echo()



Where TESCONUMBER is my Tesco telephone number, INTESCONUMBER is the international version of my tesco number (ie drop the leading 0 and add 44) and password is the password. Note the 9 after the INTTESCONUMBER. I was able to figure this out from the information about using the Australian Firefly/Freshtel service, as they are the people who provided Tesco the voip system and from looking at where Tesco's client is calling out to (gateway.tescointernetphone.com). This mostly works - I can dial out and recieve calls, but reception only works for a short period of time after I start asterisk or I reload iax2. Fortunatly, I think I've worked out why.

When asterisk registers itself with Tesco, one of the parameters that gets sent back is the length of time between registration refreshes (which lets Tesco know that I'm still online and listening). If asterisk doesn't recieve that parameter back, it assumes that it should refresh every 60 seconds. However, Tesco send back a value of 0, which gives the following little warning from asterisk:

Aug 12 21:57:19 NOTICE[3917]: sched.c:234 ast_sched_add_variable: Scheduled event in 0 ms?

A side effect of getting sent a refresh value of 0 is that Asterisk never sends out a registration refresh request, so Tesco assumes that we've gone offline. From what I've been able to gather from seaching through google and looking at the source to asterisk, there is no way to overide the setting that gets sent from the server (though I could be wrong). As such, I modified the source so that if it gets a value of 0, it changes it to 60. So far, this seems to work.

Scot's Voip Log

This blog is about my adventures with Voice Over IP. So far, I have an ancient compaq deskpro running Ubuntu Server with Asterisk installed and an el-cheapo X100p clone card.