I have a problem with my dialplan. Simply when a SIP call comes in via context from-sipgate, the call is answered and passed to the menu.
I hear the welcome message and then the call is terminated before I have a chance to press any digits.
I get on the console:
– Executing [4863581@from-sipgate:1] [color=cyan]Answer/color in new stack
– Executing [4863581@from-sipgate:2] [color=cyan]Goto/color in new stack
– Goto (menu,s,1)
– Executing [s@menu:1] [color=cyan]Playback/color in new stack
– <SIP/4863730-09dbe6f8> Playing ‘welcome’ (language ‘en’)
== Auto fallthrough, channel ‘SIP/4863730-09dbe6f8’ status is ‘UNKNOWN’
– Executing [h@menu:1] [color=cyan]Hangup/color in new stack
== Spawn extension (menu, h, 1) exited non-zero on ‘SIP/4863730-09dbe6f8’
I’m using the latest Asterisk 1.4 on CentOS 5.2
[general]
static = yes
writeprotect = yes
priorityjumping = yes
clearglobalvars = yes
[from-sipgate]
exten => h,1,Hangup
exten => 4863581,1,Answer
exten => 4863581,2,Goto(menu,s,1)
[menu]
exten => s,1,Playback(welcome)
exten => s,2,Set(TIMEOUT(digit)=10)
exten => s,3,Set(TIMEOUT(response)=60)
exten => 1,1,Goto(listen,s,1)
exten => *,1,Goto(s,1)
exten => #,1,Goto(listen,s,1)
exten => i,1,Goto(listen,s,1)
exten => t,1,Goto(listen,s,1)
exten => h,1,Hangup
[listen]
…