Any sugestions would be helpful at being able to forward it to the SIP phone if it is online and avaliable but then let that fail and drop into voicemail if it is not online or is busy.
cheers
David
-- Executing Dial("IAX2/[EMAIL PROTECTED]/3", "SIP/800|5") in new stack
Dec 21 00:15:57 NOTICE[3922]: app_dial.c:800 dial_exec: Unable to create channel of type 'SIP' (cause 3)
== Everyone is busy/congested at this time
-- Executing WaitExten("IAX2/[EMAIL PROTECTED]/3", "") in new stack
The Extensions.conf file for that section is
exten => s,1,Wait,1 exten => s,n,Answer exten => s,n,DigitTimeout,3 exten => s,n,ResponseTimeout,5 exten => s,n,Dial(SIP/800,5) exten => s,n,Waitexten exten => s,n,Playback,voicemail/default/801/unavail exten => s,n,Voicemail,801 exten => s,n,Goto,t|1
and I have in sip.conf
[800] type=friend regexten=800 username=800 secret=password callerid=800 host=dynamic ;dtmfmode=inband mailbox=800 nat=yes canreinvite=no qualify=yes disallow=all allow=gsm allow=speex allow=ilbc allow=ulaw allow=alaw _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
