This is a monthly posting giving information about the Usenet group uk.telecom.voip. If you are a new user, or even if you've been around for some time, please take a few minutes to read through the charter and bear it in mind when posting. Charter for uk.telecom.voip ===================

I have a grandcentral.com account that forwards all calls made to a DID number over to a Gizmo5 SIP number. This works great with the SPA-3102 phone configured to use the Gizmo5 service. The problem is that I'd like to use another VOIP (SIP) provider whenever I make an outgoing call. Is there a way for the SIpura to be logged onto to 2 SIP VOIP providers simultaneously, so that the phone can...

Hi, Does anyone know if it is possible to stop the Add contact thing from popping up when I make a call using SMSListo 4.01 build 476? Thanks. Wayne.

Nice bit of kit , bought from gradwell and came preconfigured for some numbers i redirected working wonderfully , very clear lines question is ! , my landline !! can i run tht thru the grandstream no sign of any socket ? dont think i can I think I can plug landline into back of router , fine Ok but still means having two phones !! anyway of running my standard landline thru the grandstream ?

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All I want is a UK (and maybe also a USA) dial-in number which forwards A-law (or ulaw in USA) audio packets from the PSTN to my asterisk server without changing the audio in any way. This means no echo cancellation. Can anyone offer such a service? I could do it myself with a group of ISDN2 lines or preferably an ISDN30 connection, but this is rather expensive.

Test. Please ignore.

Has anyone succeeded in setting up Asterisk to send caller ID with Betamax/Finarea providers? With SMSListo.com I've registered my number using the Windows application, received the confirmation call, and the app says my number is OK. On an ATA the settings to provide CLID a username=00441234567890 auth user=myusername password=mypassword

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I am trying to get presence working on a Snom 320 with OpenSER v 1.3.2. However the phone is not setting the sla parameter in the dialog Event which is required by OpenSER. However the parameter is set on this example from the Snom Wiki http://wiki.snom.com/SIP_Traces Any ideas how to enable it?

Does anyone know how you turn off the annoying Info Services? I could not care less what the world weather is going to be like. PDF manual says to turn it off on web page: SettingsServices but this page does not exist. I do have: SettingsInfo Services But this only allows me to select: Activate Info Services: YES or NO There is no button to keep this setting turned off!

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The power went out at my house. I have voice over internet. Now my GE 24 GHz phone screen readout says, "Out of Range". I have disconnected my modem, router and Lynksis adapter and phone connection then reconnected them. I still get the same readout. I have no other phone to use to say call my ISP or VOIP. What should I try to do? Please help. Earl Williams, Surrey, British Columbia

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Does anyone here use a Pluscom SIP VoIP ATA VPA-11 I bought one some while back but not used it till now. My SPA3K has gone down and I've tried this spare box. Seems to work OK except that it dies each night. Each morning I find that there is no connection and the only way to reconnect is to switch it off and on again. Every morning it's the same.

Hello, I've a customer who has two premises close to one another. On each building is a Motorola "Square-ial" supplying a microwave link for the two sites' single LAN. Q1. Is this likely to be a duplex link? Q2. If Q1 is yes, anyone got any suggestions for getting a phone

seem to be getting spam and not much else on here

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Just spotted these on eBay: http://cgi.ebay.co.uk/Brand-New-Sealed-QUARTEL-Q710-VOIP-NETWORK-SIP- PHONE_W0QQitemZ150266332064QQ but I'm damned if I can find any reviews of them, or firmware releases on the manufacturer's site . Look very similar to Aastra/Nortel handsets, presumably because they've been made in the same factory. Supports IAX as well. Anyone know anything else about them?