[asterisk-users] Experimenting- Sip dialing with Zap

[Eric "ManxPower" Wieling]

John Meksavan wrote:
> After sending the email out, I went back to change the line in 
> extensions.conf from
> 
> Dial({Zap/g0/{EXTEN:1})
> 
> to
> 
> exten => _XXX,1,Dial(Zap/g0/{EXTEN})
> 
> I am using a phone simulator to test because I do not have the physical PSTN 
> line yet. The phone simulator only allow 3 digit dialing. Now, I get this 
> message on the Asterisk CLI
> 
>     -- Executing [103 at default:1] Dial("SIP/200-006fd1a0", "Zap/g0/{EXTEN}") 
> in new stack
> [Aug 16 20:22:34] WARNING[14292]: app_dial.c:1106 dial_exec_full: Unable to 
> create channel of type 'Zap' (cause 0 - Unknown)
>   == Everyone is busy/congested at this time (1:0/0/1)
>   == Auto fallthrough, channel 'SIP/200-006fd1a0' status is 'CHANUNAVAIL'
> 

Be more careful when checking your work.

Zap is the technology, not (Zap.

The destination is ${EXTEN} not {EXTEN}.