[asterisk-users] clone X100p+dahdi dial out works only after receiving call

[Michael Higgins]

On Sat, 28 Feb 2009 21:52:46 +0200
Tzafrir Cohen <tzafrir.cohen at xorcom.com> wrote:

> On Sat, Feb 28, 2009 at 11:24:53AM -0800, Michael Higgins wrote:
> > 
> > So, tweaking configs, rebuilding this and that... restarting,
> > twiddling, it works (yeah!), but fails on re-boot to work at all.
> > Consistently, though.
> > 
> > I believe it comes down to this: I can call out only *after* I've
> > received a call.

[8<]

> >   modprobe wctc4xxp
> 
> Why? Do you have a transcoder card?

Doh! No, I suppose I don't.

> 
> >   modprobe wcfxo

[8<]

> > 
> > So, by chance, instead of ripping my hair for a bit, just to be
> > sure it's still working *at all*, I call myself:
> > 
> > starting simple switch on 'DAHDI/1-1'
> > [Feb 28 11:00:49] NOTICE[2458]: chan_dahdi.c:7125 ss_thread: Got
> > event 18 (Ring Begin)... == Starting DAHDI/1-1 at from-pstn,s,1
> > failed so falling back to exten 's' == Starting DAHDI/1-1 at
> > from-pstn,s,1 still failed so falling back to context 'default'
> 
> asterisk -rx 'dialplan show s at from-pstn'

asterisk -rx 'dialplan show s at from-pstn'
There is no existence of 'from-pstn' context
Command 'dialplan show s at from-pstn' failed.

Um... is that a clue? '-)

Can I just stick everything in 'default', or do I need different contexts?

> 
> >     -- Executing [s at default:1] Verbose("DAHDI/1-1", "1|dumb
> > answering machine") in new stack 1|dumb answering machine
> >     -- Executing [s at default:2] Answer("DAHDI/1-1", "") in new stack
> >     -- Executing [s at default:3] Playback("DAHDI/1-1",
> > "transfer,skip") in new stack -- <DAHDI/1-1> Playing
> > 'transfer.gsm' (language 'en') -- Executing [s at default:4]
> > Dial("DAHDI/1-1", "SIP/mykhyggz at 192.168.0.100,20,rt") in new stack
> > == Using SIP RTP CoS mark 5 -- Called mykhyggz at 192.168.0.100
> >     -- SIP/192.168.0.100-0827a188 is ringing
> >     -- SIP/192.168.0.100-0827a188 answered DAHDI/1-1
> >   == Spawn extension (default, s, 4) exited non-zero on 'DAHDI/1-1'
> >     -- Hungup 'DAHDI/1-1'
> > 
> > And I get my call... with success.
> > 
> > Now, I try to call out, originate at CLI again:
> > 
> > *CLI> originate DAHDI/1/5034735882 extension linphone
> >   == Starting DAHDI/1-1 at default,linphone,1 failed so falling
> > back to exten 's' -- Executing [s at default:1] Verbose("DAHDI/1-1",
> > "1|dumb answering machine") in new stack 1|dumb answering machine
> >     -- Executing [s at default:2] Answer("DAHDI/1-1", "") in new stack
> >     -- Executing [s at default:3] Playback("DAHDI/1-1",
> > "transfer,skip") in new stack -- <DAHDI/1-1> Playing
> > 'transfer.gsm' (language 'en') *CLI>     -- Executing [s at default:4]
> > Dial("DAHDI/1-1", "SIP/mykhyggz at 192.168.0.100,20,rt") in new stack
> > == Using SIP RTP CoS mark 5 -- Called mykhyggz at 192.168.0.100
> >     -- SIP/192.168.0.100-0827a700 is ringing
> >     -- SIP/192.168.0.100-0827a700 answered DAHDI/1-1
> >   == Spawn extension (default, s, 4) exited non-zero on 'DAHDI/1-1'
> >     -- Hungup 'DAHDI/1-1'
> > 

[8<]

> 
> If you configure things manually, don't also include dahdi-channels.
> If you do include it, it is probably best to include it after you set
> all the defaults in the lines below.

Okay. I removed it, and just put in the bits generated in that file, less the reference to a non-existent context line. 

Now, the DAHDI context shows as 'default'. Is that okay?

[8<]

asterisk -rx 'dialplan show s at default'
[ Context 'default' created by 'pbx_config' ]

[ Context 'default' created by 'pbx_config' ]
  's' =>            1. Verbose(1|dumb answering machine)          [pbx_config]
                    2. Answer()                                   [pbx_config]
                    3. Playback(transfer,skip)                    [pbx_config]
                    4. Dial(SIP/mykhyggz at 192.168.0.100,20,rt)     [pbx_config]
                    5. BackGround(asterisk-recording)             [pbx_config]
                    6. Voicemail(6666 at default)                    [pbx_config]
                    7. Playback(tt-weasels)                       [pbx_config]
                    8. Hangup()                                   [pbx_config]

-= 1 extension (8 priorities) in 1 context. =-


asterisk -rx 'dialplan show 6666 at default'

[ Context 'default' created by 'pbx_config' ]
  '6666' =>         1. Voicemail(6666 at default)                    [pbx_config]
                    2. Hangup()                                   [pbx_config]
  '_X.' =>          1. Dial(DAHDI/1/${EXTEN})                     [pbx_config]

-= 2 extensions (3 priorities) in 1 context. =-

... and it becomes quite clear I have NO IDEA what I'm doing. Why is that last line in the 6666 dialplan..??

Does any of this make clear why I can't *originate* a call in CLI before *receiving* a call? 

IOW, if I fix this dialplan thing, should the problem go away? Trying to eliminate other possible (hardware, driver) issues is a distraction at this point.

"originate DAHDI/1/5551212 extension linphone"... there a more specific variant on this in CLI "originate" command I could use, or something? Like to specify a context..??

Obviously I need to read up on something, just not sure what the minimum is, or where it lives. '-)

Cheers,

-- 
 |\  /|        |   |          ~ ~  
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 |    |ichael  |   |iggins    \^ /
 michael.higgins[at]evolone[dot]org