Re: [Asterisk-Users] Intervivo sip.conf?
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 intervivo, regardless of dtmfmode setting, DTMF tones are not recognised by the recipient. The same applies to receiving dtmf digits. I did mention that I never got around to making DTMF work from my home Asterisk server, but it will be possible. My guess is that there is a mis-match between the DTMF mode settings at either end, i.e. in your config and in our server config. We have a (hidden by default) config option on your control panel that allows you to specify the DTMF mode manually, which should allow us to fix this for you. Also, unless I set insecure=very (which I shouldn't need to), I get these messages when someone tries to call in: Oct 16 18:08:21 NOTICE[7175]: chan_sip.c:7162 handle_request: Failed to authenticate user xxx sip:[EMAIL PROTECTED];tag=as30592e8c where xxx is the number they're calling from. They get a busy signal. Any ideas? I'm sure we'll sort it once I've seen your config files. Cheers, Mark. p.s. If you're not keen on emailing your config files to my home address (why should you believe that I really work for InterViVo) then feel free to email them to [EMAIL PROTECTED] instead and I'll grab them from there. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk console from xinetd?
Hello Nicols, On Wed, 2004-09-08 at 14:17, Nicols Gudio 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 be honest I didn't even realise it was there. I've now read manager.txt and it does look like what I need. Thanks for the tip. Cheers, Mark. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk console from xinetd?
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= 1234
wait= no
user= root
server = /usr/sbin/asterisk
server_args = -r -n
log_on_failure += USERID
}
After adding actl to /etc/services and restarting xinetd it reports
one new service. When connecting to port 1234 on 127.0.0.1 (iptables
preventing remote hosts from accessing this service) I see the CLI prompt
repeating over and over with no line breaks.
Any idea how to prevent the looping please?
Thanks,
Mark.
p.s. Why am I doing this? We have an application that already knows
how to talk to other things via TCP sockets and we'd like to make it
talk to Asterisk too. The network between the two servers is trusted
so sending stuff clear-text isn't a problem.
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Re: [Asterisk-Users] Where can i get an UK SIP account with UK number?
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 options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CDR destination when user presses '#'
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 extensions.conf and I've tried calling ResetCDR(w) in the '#' target hoping that would cause a CDR to be written with the original number but in both cases the CDR still contains '#'. Any ideas please? Thanks, Mark. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP in the UK
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 would because I work there. :) Cheers, Mark. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Talking SIP to Vocal
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 call. A rollback to CVS of a week ago fixes the issue. I did another CVS update and rebuild last night... and outgoing SIP authentication appears to work correctly now. Did someone fix the problem? Cheers, Mark. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Talking SIP to Vocal
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 fromuser=08452416761 externip=mt104.dyndns.org nat=yes canreinvite=no reinvite=no notransfer=yes In extensions.conf I have: exten = 150,1,Dial(SIP/[EMAIL PROTECTED]) When I call 150 Asterisk sends an invite to Vocal which then asks for authentication. Asterisk sends another invite with auth details *but* the digest username is 0800800150 when it should (I think) be 08452416761. I'm using source from CVS, checked out yesterday. Calls out via IAX work fine. Calls out via SIP to Free World Dialup work fine, but then FWD doesn't ask for authentication. Is this a bug in the SIP auth code or am I misconfiged? Any ideas please? Thanks, Mark. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
