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 ===================

Hi All, Does anyone have any experience of using Verizon's VoIP service outside the US? In the light of Vonage's troubles, I am looking at alternative providers. I am particularly concerned that my US numbers should function during periods when I am in the UK. Thanks

During a call the person at the other end says my voice is very quiet and I have to speak quite loudly for them to be able to hear me clearly. Their voice is perfect at my end. I am using a Linksys SPA 3102 - what do I need to tweak please? This is for calls going out through to VOIP gateway via VoipDiscount. Thanx

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Have no landline and have a ata/router. Currently with sipdiscount and sipgate I think. Notice they now charge 1p for landline calls. I remember seeing a service that gives you 300 mins for free per week for 3 months after you top up??? If someone can remind me of this company, can you tell me what the minimum 3 monthly top up must be??? Or any other IP companies that can help, just want UK...

Can anybody help me in setting QoS in my ADSL router? Must I put the IP of my SIP provider in "Destination IP" field and my local IP in "source IP"? What is the source and the destination netmask? The router vendor didn't write any manual. :-( Thanks in advance. Router: UT-Starcom UT-300R2U Screenshot: http://img45.imageshack.us/img45/4802/ipqos1vu9.png

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 ===================

My setup consists of a SIP phone and DECT handset with integral TAM plugged into Line 1 of my SPA3k. These are then setup as extensions on Asterisk (Trixbox) with a couple of VOIP provider trunks and also the PSTN port of the SPA3k. Incoming calls on the PSTN line call the 's' extension on Asterisk and this then rings the DECT and SIP phones (setup as a group) Enough background - Down to...

Our typical VoIP installations utilise ISDN Gateways (BRI&PRI). I have a requirement to install a PBX but to interface the PSTN network using analogue lines (2 x FXO). In the past I have tried both Clipcomm and Grandstream gateways but have had limited success in getting these to work. The problems were generally related to echo, gain and the inability to detect both hangups, ringing and callerID...

All calls to my cable line come up as Witheld however calls to VOIP lines are fine. Is there something I need to change in my Fritzbox ATA?

http://www.slashgear.com/n95-voip-crippling-caught-on-camera-can-carriers-evolve-164795.php I don't imagine Orange is alone in this practice. Can this be undone by a Nokia service agent resetting the phone? thread on Truphone forum as well http://www.mobilevoipforum.org/showthread.php?t=515

Sorry to be somewhat OT but I know many of you use DECT phones plugged into the ATA. I have bought a new house (for me) and it has quite a lot of garden around it. I have a BT Freestyle 2200 system and also a Philips 1222 system. For reasons that I won't go into I need to leave the existing systems where they are. Under the same conditions I find the Philips has a better range than the BT...

Is anyone else having problems with making outgoing calls though voiptalk at the moment? Our incoming number works fine, (well the one I've tested). Their support website is down at the moment.

Hi, We've linked the servers and can call both sets of extensions etc. If I try to call out over one of the trunks on the remote server the call fails and I see NOTICE: chan_iax2.c:6793 socket_process: Rejected connect attempt from xx.xx.xx.xx, request '1571@default' does not exist.

Hi, does anyone here have experience of running the FRITZ!Box Fon ATA without a standard telephone line. The unit in question is the ATA PBX that has no ADSL 'modem'. Previous threads here indicate that it is possible to operate the FRITZ!Box Fon ATA in VoIP only mode and ignore the unit's provision for a standard 'phone line. However, yesterday I located the manual for said ATA on-line...

Hi all, I need to reset the time on my fritz, but there's no mention of it in the manual. You have to do it using the telephone keypad, I've done it once before. Does anyone know how to do it please, before I contact AVM support? Cheers

Sometime ago I remember seeing a tutorial on linking 2 asterisk together. I thought the tutorial was based around freepbx, but I can't find it any more! Can anyone point me in the correct direction? Cheers

Any suggestions for a free or cheap voip-in number in Canada (Alberta)? Either with or without a SIP account. TIA --

"* Max 300 minutes per week of free calls, measured over the last 7 days and per unique IP address." This doesn't make any sense to me. Surely they would monitor this on a per account basis. What am I missing here? -- Graham. %Profound_observation%

Anyone know what settings I should use on this, I have put my six digit username in User Name and used the correct password. I have also put registrar.voip.co.uk in the Registrar section, and left stun-server blank. However it is not working, the log is showing "Registration of Internet number XXXXX failed. Remote site reports reason for error 403". Can anyone help

anyone knows if it's possible to get a free incoming local number in US? thanks