[Asterisk-Users] Outgoing Calls via chan_capi

[Matthias Böhm]

Just found out my self :-D. Problem was a syntax change in 0.4.0pre1
version of chan_capi.
All the resources I found on the web were written with the old syntax.

Matthias


Am Donnerstag, den 02.06.2005, 17:01 +0200 schrieb Matthias BXhm:
> Hello,
> I am new to asterisk (i started to try tree days ago) and i have managed
> to setup asterisk to fit my needs so far exept one thing. 
> 
> I want to setup asterisk as an SIP <> ISDN Gateway. As I said things are
> going well so far. I can make calls from SIP phones to other SIP phones,
> and incoming calls from ISDN are forwarded to a SIP Phone correctly. But
> when I try to make a call from the SIP Phone to an ISDN Number I get
> following Error messages in the asterisk console:
> 
> ------------snip--------------
> 
>   -- Executing Dial("SIP/mboehm-fcf8", "CAPI/@123456:00174123123|30") in
> new stack
> Jun  2 15:47:37 ERROR[3376]: chan_capi.c:1181 capi_request: Syntax error
> in dialstring. read the docs!
> Jun  2 15:47:37 NOTICE[3376]: chan_capi.c:1238 capi_request: didn't find
> capi device with controller = 0 or group = 0.
> Jun  2 15:47:37 NOTICE[3376]: app_dial.c:746 dial_exec: Unable to create
> channel of type 'CAPI'
>   == Everyone is busy/congested at this time
>     -- Executing Hangup("SIP/mboehm-fcf8", "") in new stack
>   == Spawn extension (default, 001741895465, 2) exited non-zero on
> 'SIP/mboehm-fcf8'
>     -- Executing Hangup("SIP/mboehm-fcf8", "") in new stack
>   == Spawn extension (default, h, 1) exited non-zero on
> 'SIP/mboehm-fcf8'
> 
> ------------/snip-------------
> 
> Here is my capi.conf
> 
> -----------snip---------------
> 
> [general]
> nationalprefix = 0
> internationalprefix = 00
> rxgain = 0.8
> txgain = 0.8
> 
> [interfaces]
> context = default
> isdnmode=  ptmp
> msn = 123456
> incomingmsn=123456
> group = 0
> controller = 1
> softdtmf = 1
> devices = 2
> 
> -----------/snip-------------
> 
> And my extensions.conf
> 
> -----------snip--------------
> 
> [general]
> 
> static = yes
> writeprotect = no
> 
> [globals]
> 
> [default]
> 
> exten => h,1,Hangup
> 
> exten => 10,1,Dial(SIP/mboehm,30,tr)
> exten => 10,2,VoiceMail(10)
> exten => 10,3,Hangup
> 
> exten => 11,1,Dial(SIP/mboehm_note,30,tr)
> exten => 11,2,VoiceMail(11)
> exten => 11,3,Hangup
> 
> exten => 20,1,VoiceMailMain(10)
> exten => 20,2,Hangup
> 
> exten => 21,1,VoiceMailMain(11)
> exten => 21,2,Hangup
> 
> exten => 123456,1,Dial(SIP/mboehm,30,tr)
> exten => 123456,2,VoiceMail(u10)
> exten => 123456,3,Hangup
> 
> exten => _0.,1,Dial(CAPI/@123456:${EXTEN-1},30)
> exten => _0.,2,Hangup
> 
> -----------snip-------------
> 
> I am running asterisk-1.0.7 on a gentoo linux box with kernel-2.6.11 and
> an AVM Fritz Card PCI ISDN Controller. 
> I have been looking arround on all known asterisk sites but I vould not
> find anything that helped me aout. Perhaps I have better luck here.
> 
> Geetz from Northern Germany ans Sorry for my bad english
> 
> Matthias
> 
> 
> 
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