Re: [Asterisk-Users] Intervivo sip.conf?

2004-10-18 Thread Mark Turner
Hi Dave, On Sun, 17 Oct 2004, David Croft wrote: > > I have tried your config and variations on it but have the same problems. Sorry to hear that you're still having problems. If you email me your sip.conf and extensions.conf then I'd be happy to take a look. > Placing a call out using intervi

Re: [Asterisk-Users] asterisk console from xinetd?

2004-09-09 Thread Mark Turner
Hello NicolÃs, On Wed, 2004-09-08 at 14:17, NicolÃs GudiÃo wrote: > Did you try asterisk manager? You can execute all of the cli commands > and much more. Just enable it in /etc/asterisk/manager.conf and read > manager.txt in the asterisk docs directory. No, I haven't tried asterisk manager. To

[Asterisk-Users] asterisk console from xinetd?

2004-09-08 Thread Mark Turner
I'm trying to set up xinetd to run an asterisk console on a tcp port. So far I've added a file in /etc/xinetd.d/ like: service actl { disable = no socket_type = stream protocol= tcp port

Re: [Asterisk-Users] Where can i get an UK SIP account with UK number?

2004-07-16 Thread Mark Turner
Dameon D. Welch-Abernathy wrote: There are probably others. Such as www.intervivo.net. Cheers, Mark. p.s. I work there. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update opti

[Asterisk-Users] CDR destination when user presses '#'

2004-05-24 Thread Mark Turner
If '#' is pressed during a call the CDR that is written at the end of the call contains '#' in the dst / destination field rather than the number that was originally called. How do I avoid losing that original number so that I can use the CDR for billing? I've tried not having a '#' target in

Re: [Asterisk-Users] SIP in the UK

2004-05-10 Thread Mark Turner
Jeremy Bogan wrote: Does anyone know of any providers that can offer local numbers based in the UK via IAX or SIP? We're looking at getting a number based in the UK. Take a look at http://www.voip-info.org/tiki-index.php?page=VOIP%20Service%20Providers I can recommend InterViVo, but then I

Re: [Asterisk-Users] Talking SIP to Vocal

2004-05-03 Thread Mark Turner
Andres wrote: I think the username/secret items in sip.conf are busted. A quick ethereal trace shows that when placing an outbound call to another provider via SIP, * is not using the username defined during the authentication challenge, instead it uses the username of the phone placing the ca

[Asterisk-Users] Talking SIP to Vocal

2004-05-02 Thread Mark Turner
I'm trying to get Asterisk to talk SIP to Vocal and so far have only managed to get it partially working. Calls in from Vocal are working fine but outbound calls aren't. In sip.conf I have: [ivv] secret=SECRET username=08452416761 host=sip.intervivo.net