http://bugs.digium.com/bug_view_page.php?bug_id=0002905
bkw > -----Original Message----- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of C F > Sent: Sunday, December 19, 2004 8:29 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] call screening > > OK I now know what was/is worng, my SIP is wrong it doesn't give 2 way > audio, so first I'm going to fix this and then we will see. > > > On Sun, 19 Dec 2004 19:26:59 -0500, C F <[EMAIL PROTECTED]> wrote: > > Right now I'm stuck at this point: > > [default] > > exten => 1002,Macro(stdcs,1002,SIP/1002) > > > > [macro-stdcs] > > ;; arg1 exten > > ;; arg2 device > > exten => s,1,Wait(0.2) > > exten => s,2,Playback(vm-rec-name) > > exten => s,3,SetVar(SCREEN_FILE=/tmp/${CALLERIDNUM}-${EPOCH}) > > exten => s,4,Record(${SCREEN_FILE}:gsm|2|4) > > exten => s,5,Playback(pls-wait-connect-call) > > exten => s,6,Dial(${ARG2},30,gM(screen^${SCREEN_FILE})) > > exten => s,7,Voicemail(u${ARG1}) > > exten => s,8,Playback(Goodbye) > > exten => s,9,Hangup > > exten => s,107,Voicemail(b${ARG1}) > > exten => s,108,Playback(Goodbye) > > exten => s,109,Hangup > > > > [macro-screen] > > exten => s,1,Wait(0.2) > > exten => s,2,Playback(${ARG1}) > > ;1 TO ACCEPT, 2 TO REJECT, 3 TO TRANSFER > > exten => s,3,Read(ACCEPT1|custom/2) ;this file contains the phrase'you > > have an incoming call from' > > exten => s,4,Noop(${ACCEPT1}) > > exten => s,5,Gotoif($[${ACCEPT1}=1] ?50) ;connect > > exten => s,6,Gotoif($[${ACCEPT1}=2] ?30) ;reject to vm > > ;exten => s,6,Gotoif($[${ACCEPT1}=3] ?40) ;TRANSFER > > exten => s,7,Gotoif($[${ACCEPT1}=4] ?50:50) ;any thing else connect > > > > exten => s,30,SetVar(MACRO_RESULT=CONTINUE) > > exten => s,31,System(/bin/rm ${ARG1}) > > ;not yet written > > ;exten => s,40, ;ask for extension then set macro to goto that and > continue > > exten => s,50,System(/bin/rm ${ARG1}) > > > > when I dial exten 1002 I get the follwoing in the CLI: > > -- Executing Macro("SIP/1000-906f", "stdcs|1002|SIP/1002") in new stack > > -- Executing Wait("SIP/1000-906f", "0.2") in new stack > > -- Executing Playback("SIP/1000-906f", "vm-rec-name") in new stack > > -- Playing 'vm-rec-name' (language 'en') > > -- Executing SetVar("SIP/1000-906f", > > "SCREEN_FILE=/tmp/1000-1103501744") in new stack > > -- Executing Record("SIP/1000-906f", > > "/tmp/1000-1103501744:gsm|2|4") in new stack > > -- Playing 'beep' (language 'en') > > -- Executing Playback("SIP/1000-906f", "pls-wait-connect-call") in > > new stack -- Playing 'pls-wait-connect-call' (language 'en') > > -- Executing Dial("SIP/1000-906f", > > "SIP/1002|30|gM(screen^/tmp/1000-1103501744)") in new stack > > -- Called 1002 > > -- SIP/1002-1507 is ringing > > -- SIP/1002-1507 answered SIP/1000-906f > > -- Executing Wait("SIP/1001-1507", "0.2") in new stack > > -- Executing Playback("SIP/1002-1507", "/tmp/1000-1103501744") in > new stack > > -- Playing '/tmp/1000-1103501744' (language 'en') > > -- Executing Read("SIP/1002-1507", "ACCEPT1|custom/2") in new stack > > -- Playing 'custom/2' (language 'en') > > -- User entered '' > > -- Executing NoOp("SIP/1001-1507", "") in new stack > > -- Executing GotoIf("SIP/1001-1507", "=1 50") in new stack > > -- Executing GotoIf("SIP/1001-1507", "=2 30") in new stack > > -- Attempting native bridge of SIP/1000-906f and SIP/1002-1507 > > -- Executing VoiceMail("SIP/1002-906f", "u1002") in new stack > > -- Playing 'voicemail/default/1002/unavail' (language 'en') > > == Spawn extension (macro-stdcs, s, 7) exited non-zero on > > 'SIP/1000-906f' in macro 'stdcs' > > == Spawn extension (default, 1002, 1) exited non-zero on 'SIP/1000- > 906f' > > > > I have no clue why the Read doesn't work, for some reason it refuses > > to work from within this macro but works from any where else. Need > > help ASAP. > > > > > > On Sun, 19 Dec 2004 18:37:40 -0500, C F <[EMAIL PROTECTED]> wrote: > > > According to this it exists: > > > http://www.voip-info.org/wiki-Asterisk+cmd+Dial > > > However I'm testing it for the last 8 hours with no success. > > > Recompiling after reading this: > > > http://bugs.digium.com/bug_view_page.php?bug_id=0002905 > > > will post back > > > > > > > > > On Sun, 19 Dec 2004 14:46:01 -0800, Tracy R Reed > > > <[EMAIL PROTECTED]> wrote: > > > > On Sun, Dec 19, 2004 at 12:09:48PM +0200, Shoval Tomer spake thusly: > > > > > Is there a way to use asterisk for call screening? > > > > > > > > > > Meaning, a call comes in, asterisk answers with voicemail after I > don't > > > > > pickup, and the voicemail prompt + the caller's message a played > via the > > > > > sound card on asterisk. If I wan't to pick up, I do so by picking > up the > > > > > phone and dialing something. > > > > > Is it doable? > > > > > > > > I think I would try something like inviting the voicemail, the > caller, and > > > > an auto-answer (intercom) channel on your VOIP phone into a MeetMe > where > > > > your voiphone is not allowed to talk, only listen. Then you would > hear > > > > what is going on and if you wanted to talk to the person you could > join > > > > the MeetMe on a different line and talk to the person. > > > > > > > > -- > > > > Tracy Reed http://copilotcom.com > > > > This message is cryptographically signed for your protection. > > > > Info: http://copilotconsulting.com/sig > > > > > > > > > > > > _______________________________________________ > > > > 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 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 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