[Asterisk-Users] routing in extensions.conf
Joao Pereira
joao.pereira at fccn.pt
Fri Apr 22 09:03:19 MST 2005
Hello all,
Im using chan_capi to connect from a Siemens High Path to a Aterisk,
when I call from the Asterisk clients to the Siemens PBX, it works, when
I call from a Siemens client to a SIP(Asterisk) client, it doesnt work
and says this:
== Starting CAPI[contr1/930]/1 at default,930,1 failed so falling back
to exten 's'
== Starting CAPI[contr1/930]/1 at default,s,1 still failed so falling
back to context 'default'
Apr 22 16:50:27 WARNING[1149236544]: pbx.c:1877 ast_pbx_run: Channel
'CAPI[contr1/930]/1' sent into invalid extension 's' in context
'default', but no invalid handler
I think the problem is in the extensions.conf configuration, when the
Siemens calls the Asterisk, it starts ringing and nothing happens, but
what do I have to put in the extensions.conf to route the calls to the
correct SIP user?
Thanks
Joao
***************************************************
here s my capi.conf
[general]
nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8
[interfaces]
msn=12345678
incomingmsn=*
controller=1
softdtmf=1
accountcode=
context=default
;echosquelch=1
;echocancel=yes
devices=2
***************************************************
here s my extensions.conf
[general]
static=yes
writeprotect=no
[globals]
CONSOLE=Console/dsp ; Console interface for demo
TRUNK=CAPI
[default]
; SIP to SIP
exten => 100,1,Dial(SIP/joao)
exten => 101,1,Dial(SIP/encoder)
;SIP to Siemens
exten => 118,1,Dial,CAPI/12345678:b${EXTEN}|30
exten => 136,1,Dial,CAPI/12345678:b${EXTEN}|30
exten => 139,1,Dial,CAPI/12345678:b${EXTEN}|30
exten => 116,1,Dial,CAPI/12345678:b${EXTEN}|30
exten => 908,1,Dial,CAPI/12345678:b${EXTEN}|30
;Siemens to SIP
;exten => s,1,Dial(SIP/joao) this one works, but it always dial the SIP
user joao
exten => 555,1,Dial(SIP/joao) ; ok, this is the one that doesnt work,
how can I route the calls?
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