[asterisk-users] Dialling ZAP channel from analogue

[--[ UxBoD ]--]

Hi,

Asterisk Version : 1.2.15
Card : TDM11B (1 x FXO , 1 x FXS)

I have internal dialling working okay SIP->ZAP (analogue phone) and ZAP (analogue phone) -> SIP.

The problem comes when I try and make a outbound call.

Here is my extensions.conf :-

Code:
[incoming]
exten => s,1,GoToIfTime(17:00-09:00\mon-fri\*\*?outofhours|s,1)
exten => s,2,GoToIfTime(*\sat-sun\*\*?outofhours|s,1)
exten => s,3,Answer()
exten => s,4,Dial(Zap/1,20)
exten => s,5,Playback(cybermog)
exten => s,6,VoiceMail(5000 at incoming,s)
exten => s,7,Hangup()

[outofhours]
exten => s,1,Answer()
exten => s,2,Playback(cybermog)
exten => s,3,VoiceMail(5000 at incoming,s)
exten => s,4,Hangup()

[internal]
include => outbound-local
include => uri
exten => 123,1,VoiceMailMain(@incoming)
exten => 123,2,Hangup()
exten => 6000,1,Dial(${ANGELA},20)
exten => 6000,2,VoiceMail(5000 at incoming)
exten => 6000,3,Hangup()
exten => 6001,1,Dial(sip/uxbod,20)
exten => 6001,2,VoiceMail(5001 at incoming)
exten => 6001,3,Hangup()

[in]
exten => uxbod,1,Dial(sip/uxbod,20)
exten => uxbod,2,VoiceMail(5001 at incoming)
exten => uxbod,3,Hangup()

[outbound-local]
exten => _9NXXXXXX,1,Dial(${TRUNK}/${EXTEN:1})
exten => _9NXXXXXX,2,Congestion()
exten => _9NXXXXXX,102,Congestion()

[uri]
exten => _[a-z].,1,Macro(uridial,${EXTEN}@${SIPDOMAIN})
exten => _[A-Z].,1,Macro(uridial,${EXTEN}@${SIPDOMAIN})
exten => _X.,1,Macro(uridial,${EXTEN}@${SIPDOMAIN})

[macro-uridial]
exten => s,1,NoOp(Calling remote SIP peer ${ARG1})
exten => s,n,Dial(SIP/${ARG1},120,tr)
exten => s,n,Congestion()


When I try dialling 912345678 the above configuration thinks that I am dialling a SIP number, and does not bridge the call to the ZAP FXO channel.

What am I doing wrong please ?

This is the debug output :-

Code:
mailhub asterisk # asterisk -vvvvvvr
  == Parsing '/etc/asterisk/asterisk.conf': Found
  == Parsing '/etc/asterisk/extconfig.conf': Found
Asterisk 1.2.15, Copyright (C) 1999 - 2006 Digium, Inc. and others.
Created by Mark Spencer <markster at digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'show license' for details.
=========================================================================
Connected to Asterisk 1.2.15 currently running on mailhub (pid = 6327)
Verbosity is at least 6
    -- Remote UNIX connection
    -- Starting simple switch on 'Zap/1-1'
    -- Executing Macro("Zap/1-1", "uridial|9674035@") in new stack
    -- Executing NoOp("Zap/1-1", "Calling remote SIP peer 9674035@") in new stack
    -- Executing Dial("Zap/1-1", "SIP/9674035@|120|tr") in new stack
Feb 25 18:59:38 WARNING[6365]: chan_sip.c:1993 create_addr: No such host:
Feb 25 18:59:38 NOTICE[6365]: app_dial.c:1055 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination)
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing Congestion("Zap/1-1", "") in new stack
  == Spawn extension (macro-uridial, s, 3) exited non-zero on 'Zap/1-1' in macro 'uridial'
  == Spawn extension (macro-uridial, s, 3) exited non-zero on 'Zap/1-1'
    -- Hungup 'Zap/1-1'


It appears that everything is treated as a SIP call.

-- 
--[ UxBoD ]--
// PGP Key: "curl -s http://www.splatnix.net/uxbod.asc | gpg --import"
// Fingerprint: 543A E778 7F2D 98F1 3E50 9C1F F190 93E0 E8E8 0CF8
// Keyserver: www.keyserver.net Key-ID: 0xE8E80CF8
// SIP:uxbod at sip.splatnix.net

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