[Asterisk-Users] Asterisk on PPC & chan_capi issue

Patrick asterisk at puzzled.xs4all.nl
Mon Dec 12 05:11:17 MST 2005


On Fri, 2005-12-09 at 15:53 +0000, Jason Williams wrote:
> 
>         > > chan_capi registers fine:
>         > >
>         ********************************************************************** 
>         > >  [chan_capi.so] => (Common ISDN API for Asterisk)
>         > >   == This box has 1 capi controller(s).
>         > >   == Reading config for BRI1
>         > >     -- ast_capi_pvt BRI1-pseudo-D
>         (<MSN1>,<MSN2>,capi-in,0,2) (1,4,128) 
>         > >     -- ast_capi_pvt BRI1 (<MSN1>,<MSN2>,capi-in,0,2)
>         (1,4,128)
>         > >     -- ast_capi_pvt BRI1 (<MSN1>,<MSN2>,capi-in,0,2)
>         (1,4,128)
>         > >     -- listening on contr1 CIPmask = 0x1fff03ff 
>         > >   == Registered channel type 'CAPI' (Common ISDN API
>         Driver ($Revision:
>         > > 1.115 $) )
>         > >   == Registered application 'capiCommand'
>         > >   == Registered custom function VANITYNUMBER 
>         > >
>         > > Call from my GSM to a SIP phone (exten 1003) via ISDN/CAPI
>         (MSN2):
>         > >
>         **********************************************************************
>         > >   == BRI1: Incoming call '<my GSM>' -> '<MSN2>' 
>         > >
>         > >     -- Executing Macro("CAPI/BRI1/<MSN2>-0", "stdexten|
>         1003|SIP/1003")
>         > > in new stack
>         > >     -- Executing Dial("CAPI/BRI1/<MSN2>-0", "SIP/1003|10|
>         TtwW") in new 
>         > > stack
>         > > Dec  6 02:30:47 WARNING[28889]: channel.c:2494
>         ast_request: No
>         > > translator path exists for channel type SIP (native 65535)
>         to 0
>         > > Dec  6 02:30:47 NOTICE[28889]: app_dial.c:1011
>         dial_exec_full: Unable to 
>         > > create channel of type 'SIP' (cause 0 - Unknown)
>         > >   == Everyone is busy/congested at this time (1:0/0/1)
>  
> Looks like a codec problem when making calls to the SIP phone, ensure
> your sip phone has Alaw enabled in sip.conf, and supports the g711alaw
> codec. In its config

The phone has alaw enabled and this exact same setup works fine on a
i686 setup.

Regards,
Patrick



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