[Asterisk-Users] Ext matching problems

[Francisco Moreno]

Hello, sorry for this very late answer, I check the my inbox almost 4
times per hour but never saw the answer. Mind you it has almost 3 days
there :P.

Ok, 'cuz i've been playing around I just changed the sip channels' names
so now instead of "shipchan1001" and "sipchan1002" they are just the ext
number "1001" and "1002", but the dialplan is exactly the same.

here's the output when I press 0 to from any of the phones:
*CLI> Setting NAT on RTP to 0
Stopping retransmission on '3c2674758159-i7pcj36pjc2p at 192-168-0-65' of Response 1: Found
Setting NAT on RTP to 0
Check for res for 1001
Call from user '1001' is 1 out of 0
build_route: Contact hop: <sip:1001 at 192.168.0.65:5060;line=1>
    -- Executing Answer("SIP/1001-2def", "") in new stack
    -- Executing Playback("SIP/1001-2def", "fcopba1") in new stack
Ooh, format changed from unknown to ulaw
    -- Playing 'fcopba1' (language 'en')
Stopping retransmission on '3c2674758159-i7pcj36pjc2p at 192-168-0-65' of Response 2: Found
    -- Executing Hangup("SIP/1001-2def", "") in new stack
  == Spawn extension (default, 0, 3) exited non-zero on 'SIP/1001-2def'
    -- Executing Answer("SIP/1001-2def", "") in new stack
    -- Executing Playback("SIP/1001-2def", "invalid") in new stack
    -- Playing 'invalid' (language 'en')
    -- Executing Playback("SIP/1001-2def", "goodbye") in new stack
    -- Playing 'goodbye' (language 'en')
    -- Executing Hangup("SIP/1001-2def", "") in new stack
  == Spawn extension (default, h, 4) exited non-zero on 'SIP/1001-2def'
update_user_counter(1001) - decrement inUse counter
Stopping retransmission on '3c2674758159-i7pcj36pjc2p at 192-168-0-65' of Request 102: Found
---

And this one is when I press * to call the vm form any phone and
introduce the password (no matter if get in the mailbox or not when the
vm-machine stops answering my call the "pasvalide" context plays on.):
*CLI> Setting NAT on RTP to 0
Stopping retransmission on '3c2675717732-e5waws85elee at 192-168-0-65' of Response 1: Found
Setting NAT on RTP to 0
Check for res for 1001
Call from user '1001' is 1 out of 0
build_route: Contact hop: <sip:1001 at 192.168.0.65:5060;line=1>
    -- Executing VoiceMailMain("SIP/1001-7bab", "1001") in new stack
Ooh, format changed from unknown to ulaw
    -- Playing 'vm-password' (language 'en')
Stopping retransmission on '3c2675717732-e5waws85elee at 192-168-0-65' of Response 2: Found
Sending dtmf: 57 (9), at 192.168.0.65
Sending dtmf: 35 (#), at 192.168.0.65
    -- Incorrect password '9' for user '1001' (context = <any>)
Difference is 9096, ms is 1157
    -- Playing 'vm-incorrect' (language 'en')
    -- Playing 'vm-password' (language 'en')
Sending dtmf: 56 (8), at 192.168.0.65
Sending dtmf: 35 (#), at 192.168.0.65
    -- Incorrect password '8' for user '1001' (context = <any>)
Difference is 4184, ms is 543
    -- Playing 'vm-incorrect' (language 'en')
    -- Playing 'vm-password' (language 'en')
Sending dtmf: 55 (7), at 192.168.0.65
Sending dtmf: 35 (#), at 192.168.0.65
    -- Incorrect password '7' for user '1001' (context = <any>)
Difference is 4848, ms is 626
    -- Playing 'vm-incorrect' (language 'en')
    -- Playing 'vm-goodbye' (language 'en')
Locked path ''
Unlocked path ''
    -- Executing Hangup("SIP/1001-7bab", "") in new stack
  == Spawn extension (default, *, 2) exited non-zero on 'SIP/1001-7bab'
    -- Executing Answer("SIP/1001-7bab", "") in new stack
    -- Executing Playback("SIP/1001-7bab", "invalid") in new stack
    -- Playing 'invalid' (language 'en')
    -- Executing Playback("SIP/1001-7bab", "goodbye") in new stack
    -- Playing 'goodbye' (language 'en')
    -- Executing Hangup("SIP/1001-7bab", "") in new stack
  == Spawn extension (default, h, 4) exited non-zero on 'SIP/1001-7bab'
update_user_counter(1001) - decrement inUse counter
Stopping retransmission on '3c2675717732-e5waws85elee at 192-168-0-65' of Request 102: Found
---

Any idea???

Francisco.

P.S.: sorry I'm answering to "everybody" so this messages hits your
mailbox directly, it's that I took so long to answer so I'm not sure if
you alredy forgot the thread :P. Very sorry indeed. Not gonna happen
again.

Le lundi 21 mars 2005 à 16:03 -0500, C F a écrit :
> What is your CLI output?
> 
> 
> On Mon, 21 Mar 2005 15:03:14 -0400, Francisco Moreno
> <fmoreno at netcomvoice.com> wrote:
> > Hello everyone...
> > 
> > I'm trying to get up a testing pbx installation. Following instructions
> > of what've read from the handbook and from asterisk's wiki, I wrote the
> > dial plan as follows:
> > [general]
> > ;
> > ;
> > static = yes
> > ;[globals]
> > ;
> > 
> > [default]
> > ;
> > exten => 0,1,Answer()
> > exten => 0,2,Playback(fcopba1)
> > exten => 0,3,Hangup()
> > exten => *0,1,Answer()
> > exten => *0,2,Record(fcopba1:gsm)
> > exten => *0,3,Playback(fcopba1)
> > exten => *0,4,Hangup()
> > include => extentions
> > include => pasvalide
> > 
> > [extentions]
> > exten => 1001,1,Dial(SIP/sipchan1001,10)
> > exten => 1001,2,Voicemail(u1001)
> > exten => 1001,3,Hangup()
> > exten => 1002,1,Dial(SIP/sipchan1002,10)
> > exten => 1002,2,Voicemail(u1002)
> > exten => 1002,3,Hangup()
> > exten => *,1,VoicemailMain(${CALLERIDNUM})
> > ;exten => *,1,VoicemailMain()
> > exten => *,2,Hangup()
> > 
> > [pasvalide]
> > exten => _.,1,Answer()
> > exten => _.,2,Playback(invalid)
> > exten => _.,3,Playback(goodbye)
> > exten => _.,4,Hangup()
> > .............