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 guys (and girls?), I had a working config few days ago, but I messed up with my zaptel/zapata configs and can't get CID working. I've connected regular phone and it is showing CID so the line is okay. My log is telling me: Jun 22 10:47:39 VERBOSE logger.c: == Starting post polarity CID detection on channel 2 Jun 22 10:47:39 VERBOSE logger.c: -- Starting simple switch on

wrote: How do you restore a Sipura 2100 device to its factory original settings? I currently have it programmed to my own VOIP providers and am looking to sell it on Ebay so I want to wipe those settings. I have looked for a reset button on the unit but to no avail, is there a web screen or phone number sequence which will reset the unit to its original...

Hi all, I have a Quintum A800. I know how to configure it to do 1 DIRECT mode (gtw-to-gtw) call using CLID (111) and a prefix (123123) on the DNIS to IP 10.10.10.10: (dial with leading 00) pbxtg 1 huntpubldn 111 cid 2 exit

I seem to have broken my asterisk voicemail! It does not record anything, it usually it records in three formats i think? When watching in my CLI it only tries wav49, but then fails. So it askes you to leave message, then dont let it record, it just asks if it should be saved reviewed etc. So somehow i've broke whatever it does when recording.. Any ideas?

Hi all, I am trying to port voice recognition software to the company's server so that users can use the technology in a browser instead of in a downloaded application on the user's computer. What I need is a server that can accept VoIP data (through SIP channels, I'm guessing), process that data, and then return information to the user's computer or browser. I'm a bit of a noob in these sorts...

I have quite a lot of trunks from the same provider set on my asterisk @ home box. When a call comes in on any of these, they always come in on same 'channel' which is one of the providers accounts SIP ID's. I think in need to sort out the Incoming Settings on the trunks? Not sure what i should have in this!

http://signup.voip.demon.net/demon/products/voice/voiceoverbroadband/index.php

over the last few days, when making a call on my vonage line, upon hanging up the call has hung up at my end and I am enable to make another call, but the call at the other end has not been " hung up"! this is the first time I have had any problems with Vonage and quite impressed by their service. The fault will clear if i disconnect the PaP2 box for 2 mins or it clears itself after about...

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Hi, Has anyone put 2x TDM04B in a PC to give (upto) 8 analogue lines? Would it be ok? I imagine it would but it would be great if someone here had actually done it. thanks BW

hi If a customer migrates to ADSL MAX BT enable interleaving, which, I understand can affect latency on the line, potentially adding upto 50ms. I've got a pair of bonded MAX lines and my latency is fine, but I was wondering if anyone else was experiencing issues? thanks peter

This may not be the right place for this question but I'll try anyway.... My VOIP SP offers me a local code number for incoming calls. Will I get a BT directory entry with that number? TIA, JohnK

Hi, Anyone know of a serviced communications contact for voipcheap.co.uk (preferably customer service or accounts department). The online web mail communication form does not seem to attract a very swift response (if any) from voipcheap.co.uk. Many thanks.

I am looking to purchase a Linksys SPA3000. Which would be the best supplier to go for. I am bascially looking for an out of the box service that can give me cheap uk calls as well as option of UK lanline numbers, which I am able to choose? Any advice? Sid

Just looking for some handset feedback. I started on BT101 / 102 - Great for learning, wouldn't use (again) commercially. :-( I moved to AT-320 - Better sound than 101/102 but still a cheap feel to them. IAX2 firmware a nice option. I am considering Aastra 9112i for next installation, whats the thoughts on

Have a Linksys SPA-3000 Line one to VoIP provider works ok PSTN line set to VoIP provider 2 and can dial out calls ok SPA-3000 is also plugged into the PSTN and can receive calls OK and rings through to Line 1 The problem comes when I try and dial out on the PSTN which gets a tone

When I phone a landline from my Sipgate line my number doesn't come up on their caller display. When I phone a mobile the number is displayed. Any ideas why this would be? In my router I have an option for caller ID modulation this is currently set to CID_V23 but I have the option of changing it to CID_BELL202. Will changing this make any difference? Sorry for being thick but I'm fairly new...

Hi there, Any other Freecall users on here? what do you think of the service? What would you like to see added to the service? Bob

I've just got my Sipura SPA3000 so getting to grips with as as fast as I can. I've got Sipgate setup on line 1 which seems to be working fine for calls in and out. I'm not trying to configure Voipbuster as gateway 1. My current settings are : Line 1 tab settings Gateway 1 : GW1 NAT Mapping Enable: Yes GW1 Auth ID: username GW1 Password: password

Hmm looks like my credit has gone before I had a chance to top it up. Seems they are using a time zone of UK +1 Bummer - it was only a euro or two but it is the principle. Still I reckon I've well had my 11.75E worth and they seem to be about the only voip co that has a *reliable* call divert to a mobile from the allocated geographic no