Transition to a new line in dialplan not working :?

[system]

Hello.
transition to a new line in dialplan not working
=-=
Asterisk 1.8.7.0

extensions.conf

[general]
static=yes
writeprotect=no
autofallthrough=no
extenpatternmatchnew=no
clearglobalvars=no
userscontext=default

Dialplan

[quote]exten => 110,1,Dial(SIP/110,6,m,tT)
exten => t,n,Dial(SIP/gbmarkt/79299917867/,m,)
exten => t,n,Hangup[/quote]
=-=
My dial with 1111 to 110.

asteriskCLI>
== Using UDPTL CoS mark 5
== Using SIP RTP CoS mark 5
– Executing [110@outgoing-dial:1] Dial(“SIP/1111-0000001e”, “SIP/110,6,m,tT”) in new stack
== Using UDPTL CoS mark 5
== Using SIP RTP CoS mark 5
– Called SIP/110
– Started music on hold, class ‘default’, on SIP/1111-0000001e
– SIP/110-0000001f is ringing
asteriskCLI>
asteriskCLI>
asteriskCLI>
asterisk*CLI>
– Nobody picked up in 6000 ms
– Stopped music on hold on SIP/1111-0000001e
– Executing [110@outgoing-dial:2] Hangup(“SIP/1111-0000001e”, “”) in new stack
== Spawn extension (outgoing-dial, 110, 2) exited non-zero on ‘SIP/1111-0000001e’

that the global option is responsible for it in Asterisk 1.8?

[david55]

I believe the t extension is for incomplete numbers. For timed out dialing you need to check the DIALSTATUS value after the Dial and act accordingly.