[asterisk-users] Experimenting- Sip dialing with Zap

[John Meksavan]

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'

MY sip.conf file looks like this.

sip.conf
[general]
context=default                 ; Default context for incoming calls
bindport=5060                   ; UDP Port to bind to (SIP standard port is 
5060)
bindaddr=0.0.0.0                ; IP address to bind to (0.0.0.0 binds to 
all)
srvlookup=yes                   ; Enable DNS SRV lookups on outbound calls
tos=lowdelay                    ; 
lowdelay,throughput,reliability,mincost,none
;externip = 200.201.202.203     ; Address that we're going to put in 
outbound SIP messages
localnet=192.168.0.0/255.255.0.0; All RFC 1918 addresses are local networks
;callerid=Delta Mobile Software <8478846728>

; The Yoda VoIP wired phone drops audio when the codec is switched from
; ulaw to gsm after a ringback message is sent.  As a workaround, only
; allow ulaw and alaw.

disallow=all
;allow=all

;mike's settings
allow=ulaw
allow=alaw
#include <sip.generated>

sip.generated
[200]
username=200
secret=yeengohh
type=friend
context=default
mailbox=200
callerid=200 GenNum <8478846750>
host=dynamic
nat=no
canreinvite=no




>From: "James FitzGibbon" <james.fitzgibbon at gmail.com>
>Reply-To: Asterisk Users Mailing List - Non-Commercial 
>Discussion<asterisk-users at lists.digium.com>
>To: "Asterisk Users Mailing List - Non-Commercial 
>Discussion"<asterisk-users at lists.digium.com>
>Subject: Re: [asterisk-users] Experimenting- Sip dialing with Zap
>Date: Thu, 16 Aug 2007 12:49:16 -0400
>
>On 8/16/07, John Meksavan <jmeksavan at hotmail.com> wrote:
> >
> > CLI.  What am I doing wrong? Thanks in advance.
>
>
>The channel spec you need to use is:
>
>Dial(Zap/g0/${EXTEN:1})
>
>not
>
>Dial({Zap/g0/{EXTEN:1})
>
>Though bear in mind that the :1 is removing the first char of your
>extension, so if you dial '123' on your Linksys, you'll dial '23' out your
>analog line, which is unlikely to be what you want to do if said line is
>connected to the PSTN.  It's more typical to see something like
>
>exten => _9NXXNXXXXXX,1,Dial(Zap/g0/${EXTEN:1})
>
>which matches a 10 digit local number prefixed by nine, but removes the
>leading 9 (using :1) because it's not needed (or wanted) by the telco.
>
>--
>j.


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