thr3ads.net - asterisk users - [asterisk-users] clone X100p+dahdi dial out works only after receiving call [Feb 2009]

If this information is useful, please help other people find it:
Share via:

Michael Higgins

2009-Feb-28 19:24 UTC

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

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.

So, cold boot. Then:

  modprobe dahdi
  modprobe wctc4xxp
  modprobe wcfxo

dahdi: Telephony Interface Registered on major 196
dahdi: Version: 2.1.0.3
dahdi_transcode: Loaded.
ACPI: PCI Interrupt 0000:00:06.0[A] -> Link [LNKB] -> GSI 11 (level, low)
-> IRQ 11
Found a Wildcard FXO: Wildcard X100P

 cat /proc/dahdi/1 
Span 1: WCFXO/0 "Wildcard X100P Board 1" (MASTER) 

	   1 WCFXO/0/0 

Looks good so far. I think. Don't really know what the strings represent
entirely.

 # /etc/init.d/dahdi start
 * Starting DAHDI ...                                                    


Start asterisk:
sudo -u asterisk asterisk -cvvv

*CLI> dahdi show status
Description                              Alarms  IRQ    bpviol CRC4   Fra Codi
Options  LBO
Wildcard X100P Board 1                   OK      0      0      0      CAS Unk 
YEL      0 db (CSU)/0-133 feet (DSX-1)

*CLI> dahdi show channel 1
Channel: 1
File Descriptor: 10
Span: 1
Extension: 
Dialing: no
Context: from-pstn
Caller ID: 
Calling TON: 0
Caller ID name: 
Mailbox: none
Destroy: 0
InAlarm: 0
Signalling Type: FXS Kewlstart
Radio: 0
Owner: <None>
Real: <None>
Callwait: <None>
Threeway: <None>
Confno: -1
Propagated Conference: -1
Real in conference: 0
DSP: no
Busy Detection: no
TDD: no
Relax DTMF: no
Dialing/CallwaitCAS: 0/0
Default law: ulaw
Fax Handled: no
Pulse phone: no
DND: no
Echo Cancellation:
        1 taps
        (unless TDM bridged) currently OFF
Actual Confinfo: Num/0, Mode/0x0000
Actual Confmute: No
Hookstate (FXS only): Onhook

So, all is good. First test is to see if I can originate a call from CLI:

*CLI> originate DAHDI/1/5034735882 extension linphone
*CLI> [Feb 28 10:59:48] NOTICE[2401]: channel.c:3316 __ast_request_and_dial:
Unable to request channel DAHDI/1/5034735882

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'
    -- 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'


So, obviously it works... sort of. I'm assuming that, since I don't
really *know* what I'm doing, someone else who *does* can probably point out
the missing or incorrect part of my configuration.

(Meanwhile, I'll see about IRQ status... seems a possible culprit, now that
I think about it more.)

Anyway, here's the bits of:
./chan_dahdi.conf
./dahdi-channels.conf
./extensions.conf

[trunkgroups]
[channels]
#include /etc/asterisk/dahdi-channels.conf
signalling=fxs_ks
usecallerid=yes
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=no
echocancelwhenbridged=no
group=1
callgroup=1
pickupgroup=1
immediate=yes
ringtimeout=8000
signalling=fxs_ks
callerid=asreceived

group=0
context=from-pstn
channel => 1

[general]
static=yes
writeprotect=no
clearglobalvars=no
[globals]
CONSOLE=Console/dsp				
IAXINFO=guest					
TRUNK=DAHDI/g1					
TRUNKMSD=0

[default]
exten => 1205,1,Wait(2)
exten => 1205,2,Record(/tmp/asterisk-recording:gsm)
exten => 1205,3,Hangup
exten => s,1,Verbose(1|dumb answering machine) 
exten => s,n,Answer() 
exten => s,n,Playback(transfer,skip)		
exten => s,n,Dial(SIP/mykhyggz at 192.168.0.100,20,rt) 
exten => s,n,BackGround(asterisk-recording)
exten => s,n,Voicemail(6666 at default) 
exten => s,n,Playback(tt-weasels) 
exten => s,n,Hangup() 
exten => 4567,1,Dial(SIP/mykhyggz at 192.168.0.100,20,rt)
exten => _X.,1,Dial(DAHDI/1/${EXTEN})
exten => 3456,1,Dial(SIP/linphone,20,rt)
exten => 6666,1,Voicemail(6666 at default) 
exten => 6666,n,Hangup() 

Also, 

*CLI> dahdi restart
 Destroying channels and reloading DAHDI configuration.
       > Initial softhangup of all DAHDI channels complete.
       > Final softhangup of all DAHDI channels complete.
  == Unregistered channel 1
  == Parsing '/etc/asterisk/chan_dahdi.conf':   == Found
  == Parsing '/etc/asterisk/dahdi-channels.conf':   == Found
  == Parsing '/etc/asterisk/users.conf':   == Found
    -- Reconfigured channel 1, FXS Kewlstart signalling

but, after a cold boot and restart:

 dahdi restart
 Destroying channels and reloading DAHDI configuration.
  == Unregistered channel -2
  == Unregistered channel 1
  == Parsing '/etc/asterisk/chan_dahdi.conf':   == Found
  == Parsing '/etc/asterisk/dahdi-channels.conf':   == Found
  == Parsing '/etc/asterisk/users.conf':   == Found
    -- Reconfigured channel 1, FXS Kewlstart signalling
*CLI> 


What the heck is channel -2, I wonder?

. . .

Finally... do I *really* need to cold boot in order to re-init this card
successfully? Or is there some known sure way to get it initialized truly
'from scratch'? It seems *so* wrong to boot unless I've rebuilt the
kernel.

Thanks for any help or suggestions to fix this problem.

Cheers,

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

Tzafrir Cohen

2009-Feb-28 19:52 UTC

head link

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

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.
> 
> So, cold boot. Then:
> 
>   modprobe dahdi
>   modprobe wctc4xxp
Why? Do you have a transcoder card?
>   modprobe wcfxo
> 
> dahdi: Telephony Interface Registered on major 196
> dahdi: Version: 2.1.0.3
> dahdi_transcode: Loaded.
> ACPI: PCI Interrupt 0000:00:06.0[A] -> Link [LNKB] -> GSI 11 (level,
low) -> IRQ 11
> Found a Wildcard FXO: Wildcard X100P
> 
>  cat /proc/dahdi/1 
> Span 1: WCFXO/0 "Wildcard X100P Board 1" (MASTER) 
> 
> 	   1 WCFXO/0/0 
> 
> Looks good so far. I think. Don't really know what the strings
represent entirely.
> 
>  # /etc/init.d/dahdi start
>  * Starting DAHDI ...                                                    
> 
> 
> Start asterisk:
> sudo -u asterisk asterisk -cvvv
> 
> *CLI> dahdi show status
> Description                              Alarms  IRQ    bpviol CRC4   Fra
Codi Options  LBO
> Wildcard X100P Board 1                   OK      0      0      0      CAS
Unk  YEL      0 db (CSU)/0-133 feet (DSX-1)
> 
> *CLI> dahdi show channel 1
> Channel: 1
> File Descriptor: 10
> Span: 1
> Extension: 
> Dialing: no
> Context: from-pstn
> Caller ID: 
> Calling TON: 0
> Caller ID name: 
> Mailbox: none
> Destroy: 0
> InAlarm: 0
> Signalling Type: FXS Kewlstart
> Radio: 0
> Owner: <None>
> Real: <None>
> Callwait: <None>
> Threeway: <None>
> Confno: -1
> Propagated Conference: -1
> Real in conference: 0
> DSP: no
> Busy Detection: no
> TDD: no
> Relax DTMF: no
> Dialing/CallwaitCAS: 0/0
> Default law: ulaw
> Fax Handled: no
> Pulse phone: no
> DND: no
> Echo Cancellation:
>         1 taps
>         (unless TDM bridged) currently OFF
> Actual Confinfo: Num/0, Mode/0x0000
> Actual Confmute: No
> Hookstate (FXS only): Onhook
> 
> So, all is good. First test is to see if I can originate a call from CLI:
> 
> *CLI> originate DAHDI/1/5034735882 extension linphone
> *CLI> [Feb 28 10:59:48] NOTICE[2401]: channel.c:3316
__ast_request_and_dial: Unable to request channel DAHDI/1/5034735882
> 
> 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'
>     -- 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'
> 
> 
> So, obviously it works... sort of. I'm assuming that, since I don't
really *know* what I'm doing, someone else who *does* can probably point out
the missing or incorrect part of my configuration.
> 
> (Meanwhile, I'll see about IRQ status... seems a possible culprit, now
that I think about it more.)
> 
> Anyway, here's the bits of:
> ./chan_dahdi.conf
> ./dahdi-channels.conf
> ./extensions.conf
> 
> [trunkgroups]
> [channels]
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.
> #include /etc/asterisk/dahdi-channels.conf
> signalling=fxs_ks
> usecallerid=yes
> callwaiting=yes
> usecallingpres=yes
> callwaitingcallerid=yes
> threewaycalling=yes
> transfer=yes
> canpark=yes
> cancallforward=yes
> callreturn=yes
> echocancel=no
> echocancelwhenbridged=no
> group=1
> callgroup=1
> pickupgroup=1
> immediate=yes
> ringtimeout=8000
> signalling=fxs_ks
> callerid=asreceived
> 
> group=0
> context=from-pstn
> channel => 1
> 
> [general]
> static=yes
> writeprotect=no
> clearglobalvars=no
> [globals]
> CONSOLE=Console/dsp				
> IAXINFO=guest					
> TRUNK=DAHDI/g1					
> TRUNKMSD=0
> 
> [default]
> exten => 1205,1,Wait(2)
> exten => 1205,2,Record(/tmp/asterisk-recording:gsm)
> exten => 1205,3,Hangup
> exten => s,1,Verbose(1|dumb answering machine) 
> exten => s,n,Answer() 
> exten => s,n,Playback(transfer,skip)		
> exten => s,n,Dial(SIP/mykhyggz at 192.168.0.100,20,rt) 
> exten => s,n,BackGround(asterisk-recording)
> exten => s,n,Voicemail(6666 at default) 
> exten => s,n,Playback(tt-weasels) 
> exten => s,n,Hangup() 
> exten => 4567,1,Dial(SIP/mykhyggz at 192.168.0.100,20,rt)
> exten => _X.,1,Dial(DAHDI/1/${EXTEN})
> exten => 3456,1,Dial(SIP/linphone,20,rt)
> exten => 6666,1,Voicemail(6666 at default) 
> exten => 6666,n,Hangup() 
> 
> Also, 
> 
> *CLI> dahdi restart
>  Destroying channels and reloading DAHDI configuration.
>        > Initial softhangup of all DAHDI channels complete.
>        > Final softhangup of all DAHDI channels complete.
>   == Unregistered channel 1
>   == Parsing '/etc/asterisk/chan_dahdi.conf':   == Found
>   == Parsing '/etc/asterisk/dahdi-channels.conf':   == Found
>   == Parsing '/etc/asterisk/users.conf':   == Found
>     -- Reconfigured channel 1, FXS Kewlstart signalling
> 
> but, after a cold boot and restart:
> 
>  dahdi restart
>  Destroying channels and reloading DAHDI configuration.
>   == Unregistered channel -2
>   == Unregistered channel 1
>   == Parsing '/etc/asterisk/chan_dahdi.conf':   == Found
>   == Parsing '/etc/asterisk/dahdi-channels.conf':   == Found
>   == Parsing '/etc/asterisk/users.conf':   == Found
>     -- Reconfigured channel 1, FXS Kewlstart signalling
> *CLI> 
> 
> 
> What the heck is channel -2, I wonder?
> 
> . . .
> 
> Finally... do I *really* need to cold boot in order to re-init this card
successfully? Or is there some known sure way to get it initialized truly
'from scratch'? It seems *so* wrong to boot unless I've rebuilt the
kernel.
> 
> Thanks for any help or suggestions to fix this problem.
> 
> Cheers,
> 
> -- 
>  |\  /|        |   |          ~ ~  
>  | \/ |        |---|          `|` ?
>  |    |ichael  |   |iggins    \^ /
>  michael.higgins[at]evolone[dot]org
> 
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
-- 
               Tzafrir Cohen
icq#16849755              jabber:tzafrir.cohen at xorcom.com
+972-50-7952406           mailto:tzafrir.cohen at xorcom.com
http://www.xorcom.com  iax:guest at local.xorcom.com/tzafrir

Tzafrir Cohen

2009-Mar-03 10:04 UTC

head link

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

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.
http://bugs.digium.com/view.php?id=14577 ?

-- 
               Tzafrir Cohen
icq#16849755              jabber:tzafrir.cohen at xorcom.com
+972-50-7952406           mailto:tzafrir.cohen at xorcom.com
http://www.xorcom.com  iax:guest at local.xorcom.com/tzafrir

Reasonably Related Threads

Search for more maybe matching threads

asterisk users - Feb 2009 - clone X100p+dahdi dial out works only after receiving call

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

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

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

Reasonably Related Threads