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...SHA1 Hi all! I am trying to redirect to a local extension the incoming calls that I receive to an account which I have in iptel.org, but when receiving I'm obtaining this error: alderamin*CLI> -- Executing [300 at from-internal:1] Dial("SIP/danib-089f8820", "SIP/300|30|tTrm") in new stack [Feb 19 19:22:50] WARNING[19254]: app_dial.c:1237 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) [Feb 19 19:23:00] WARNING[19254]: pbx.c:2529 __ast_pbx_run: Timeout, but no rule 't...

...meIAXBox.com TRUNK3=SIP/bah@iconnechre.com ;key associating extension/user to a channel via a DBGet EXT1=1Zap ;see Incoming DBGet Lookup uses this family/key eg. CallFwd/1Zap EXT2=2Zap ; ect ; SPEEDIAL=megacontext ;context for call fwds when using extensions [exts] exten => 11,1,Dial(Zap/1,20,tTrm) ;dial ext on zap channel 2 exten => 12,1,Dial(Zap/2,20,tTrm) ;dial ext on zap channel 2 exten => 13,1,Dial(Zap/3,20,tTrm) ;dial ext on zap channel 3 ;... [remoteexts] exten => 1001,1,Dial(IAX/guest@myBuddy.org) ;some iax friend hosts exten => 1002,1,Dial(${TRUNK1}/12505552323) ;some P...

Is there any way to have users be able to turn on or off call forwarding at the asterisk server, so they can configure their own forwarding number and enable/disable it? Hopefully, with the added benefit that it will remain on between server reloads and restarts? I have written a hack -- a AGI script to do various checking, and if the destination is "ok" set a database variable

Hi, sorry if this is a newbie question, but in fact I am sort of a newbie. Is there a way of connecting two answered and active voice channels together in an AGI script for some time, having the two parties talk to each other, at the same time have asterisk or the AGI script listen for DTMF tones on both channels and react to certain tones, i.e. disconnecting the two channels on reception of

...default) exten => *300,n,WaitMusicOnHold(2000) exten => *300,n,Hangup ; Dial-by-name directory exten => *400,1,Directory(voicemail,from-internal) ;----------------------------------- [from-pstn] ; incoming calls from FXO port are directed to this context exten => s,1,Dial(DAHDI/2,15,tTrm) exten => s,n,Background(if-u-know-ext-dial) ; Dial known extension exten => s,n,WaitExten() include => from-internal - ------------------------------------------------------------------------- Although internally it works as I had thought in such a way that Asterisk derives to the voic...