[asterisk-users] Test

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Re: [asterisk-users] invite to conference by a call file

Here I'm using the "Page" application to make a conference call "on the fly". [office] exten => ,1,Dial(SIP/desk2,150)    same => n,Hangup() exten => ,1,Dial(SIP/desk3,150)    same => n,Hangup() exten => ,1,Dial(SIP/desk4,150)    same => n,Hangup() exten =>

Re: [asterisk-users] invite to conference by a call file

You would make the call file the same way you are now. 100 was the conf room ID. Have a look at the documentation how to do it. Also take a look at the default settings in confbridge.conf voice1*CLI> core show application ConfBridge -= Info about application 'ConfBridge' =- [Synopsis]

Re: [asterisk-users] invite to conference by a call file

thanks a lot for the reply. [call-file-test] Exten => 10,1,Answer same => ConfBridge(100) i assume 100 is the conference room, correct? where do i write the SIP numbers to invite(internal or external)? what about the PIN? On Tue, Mar 20, 2018 at 4:41 PM, Dovid Bender

Re: [asterisk-users] invite to conference by a call file

Atux, This should work: [call-file-test] Exten => 10,1,Answer same => ConfBridge(100) On Tue, Mar 20, 2018 at 10:34 AM, Atux Atux wrote: > Hi. in my system i have a conference room where someone can call it eg 698 > dial the PIN eg 1234 and enter the room as a user. The

[asterisk-users] invite to conference by a call file

Hi. in my system i have a conference room where someone can call it eg 698 dial the PIN eg 1234 and enter the room as a user. The admin enters in through a different number and PIN. I would like to have a call file and call all participants eg 610-619 at certain time of the day and give them

Re: [asterisk-users] Is 100 trying mandatory? Can asterisk answer with 180 without prior 100 trying?

Hi Tryba > A (very) dirty workaround would be to drop these packets with iptables > (assuming Linux as OS), something like: > > iptables -t raw -I OUTPUT -p udp -d ipaddrofpbx -m string --algo bm > --from 0 --to 32 --string "SIP/2.0 100 " -j DROP > > Don't try it with TCP :) :-) Indeed, this

Re: [asterisk-users] Is 100 trying mandatory? Can asterisk answer with 180 without prior 100 trying?

On Mon, Mar 19, 2018 at 12:59:47PM -0300, Joshua Colp wrote: > > To try to reproduce the problem with our SBC, is there a way to tell > > the asterisk, preferably PJSIP, to directly answer with 180 ringing > > without prior 100 trying? > > The PJSIP channel driver has no option or ability to do