Re: [Asterisk-Users] routing in extensions.conf

Stefan Helbing sth at weblab.de
Fri Apr 22 11:45:46 MST 2005


Hello Joao,

first I suggest you set an "context" string in capi.conf to lead incoming calls into a special context to give you more flexibility (in my opinion), e.g.
context=siemens
For this you need a line [siemens] in your extensions.conf.

Then (and also in the case you use the default context for everything) you need the necessary lines in extensions.conf.
If you call the number 930 from siemens to asterisk you need a line like 
exten => 930,1,DoWhatEverYouWantToDo
This line currently is missing therefor the fallback of asterisk to an "s" extensions. If you want to catch this, too (what I would recommend), you need an additional line
exten => s,1,DoStandardThings

Of course, this is only the minimum, there are much more possibilities (especially if you want to do more than one thing in an extension).

Bye
Stefan

sth>======Originalnachricht======
sth>Von: "Joao Pereira" <joao.pereira at fccn.pt>
sth>Datum: 2005-04-22 18:25:17
sth>An: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users at lists.digium.com>
sth>Betreff: [Asterisk-Users] routing in extensions.conf
sth>
sth>Hello all,
sth>Im using chan_capi to connect from a Siemens High Path to a Aterisk, 
sth>when I call from the Asterisk clients to the Siemens PBX, it works, when 
sth>I call from a Siemens client to a SIP(Asterisk) client, it doesnt work 
sth>and says this:
sth>
sth>  == Starting CAPI[contr1/930]/1 at default,930,1 failed so falling back 
sth>to exten 's'
sth>  == Starting CAPI[contr1/930]/1 at default,s,1 still failed so falling 
sth>back to context 'default'
sth>Apr 22 16:50:27 WARNING[1149236544]: pbx.c:1877 ast_pbx_run: Channel 
sth>'CAPI[contr1/930]/1' sent into invalid extension 's' in context 
sth>'default', but no invalid handler
sth>
sth>I think the problem is in the extensions.conf configuration, when the 
sth>Siemens calls the Asterisk, it starts ringing and nothing happens, but 
sth>what do I have to put in the extensions.conf  to route the calls to the 
sth>correct SIP user?
sth>Thanks
sth>Joao
sth>
sth>***************************************************
sth>here s my capi.conf
sth>
sth>[general]
sth>nationalprefix=0
sth>internationalprefix=00
sth>rxgain=0.8
sth>txgain=0.8
sth>
sth>[interfaces]
sth>msn=12345678
sth>incomingmsn=*
sth>controller=1
sth>softdtmf=1
sth>accountcode=
sth>context=default
sth>;echosquelch=1
sth>;echocancel=yes
sth>devices=2
sth>
sth>
sth>***************************************************
sth>here s my extensions.conf
sth>
sth>[general]
sth>static=yes
sth>writeprotect=no
sth>
sth>[globals]
sth>CONSOLE=Console/dsp                             ; Console interface for demo
sth>TRUNK=CAPI
sth>
sth>[default]
sth>
sth>; SIP to SIP
sth>exten => 100,1,Dial(SIP/joao)
sth>exten => 101,1,Dial(SIP/encoder)
sth>
sth>;SIP to Siemens
sth>exten => 118,1,Dial,CAPI/12345678:b${EXTEN}|30
sth>exten => 136,1,Dial,CAPI/12345678:b${EXTEN}|30
sth>exten => 139,1,Dial,CAPI/12345678:b${EXTEN}|30
sth>exten => 116,1,Dial,CAPI/12345678:b${EXTEN}|30
sth>exten => 908,1,Dial,CAPI/12345678:b${EXTEN}|30
sth>
sth>;Siemens to SIP
sth>;exten => s,1,Dial(SIP/joao)  this one works, but it always dial the SIP 
sth>user joao
sth>
sth>exten => 555,1,Dial(SIP/joao) ; ok, this is the one that doesnt work, 
sth>how can I route the calls?
sth>
sth>
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sth>



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