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

2005-04-26 Thread Joao Pereira 
Thanks Stefan, you rule...
now, tell me just one more thing please,
I putted in capi.conf :
msn=12345678
incomingmsn=*
controller=1
softdtmf=1
accountcode=
context=siemens
devices=2
and in extension.conf :
[siemens]
exten = 930,1,Dial(SIP/joao)
but this means that when 930 is dialed, user joao always receives the 
calls,
but I have 10 SIP users , and I whant that, after 930 have been dialed, 
to dial one more number to refer to each of the SIP users. How do I put 
it in extensions.conf ?
Thanks
Joao



Stefan Helbing wrote:
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==
sthVon: Joao Pereira [EMAIL PROTECTED]
sthDatum: 2005-04-22 18:25:17
sthAn: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com
sthBetreff: [Asterisk-Users] routing in extensions.conf
sth
sthHello all,
sthIm using chan_capi to connect from a Siemens High Path to a Aterisk, 
sthwhen I call from the Asterisk clients to the Siemens PBX, it works, when 
sthI call from a Siemens client to a SIP(Asterisk) client, it doesnt work 
sthand says this:
sth
sth  == Starting CAPI[contr1/930]/1 at default,930,1 failed so falling back 
sthto exten 's'
sth  == Starting CAPI[contr1/930]/1 at default,s,1 still failed so falling 
sthback to context 'default'
sthApr 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
sthI think the problem is in the extensions.conf configuration, when the 
sthSiemens calls the Asterisk, it starts ringing and nothing happens, but 
sthwhat do I have to put in the extensions.conf  to route the calls to the 
sthcorrect SIP user?
sthThanks
sthJoao
sth
sth***
sthhere s my capi.conf
sth
sth[general]
sthnationalprefix=0
sthinternationalprefix=00
sthrxgain=0.8
sthtxgain=0.8
sth
sth[interfaces]
sthmsn=12345678
sthincomingmsn=*
sthcontroller=1
sthsoftdtmf=1
sthaccountcode=
sthcontext=default
sth;echosquelch=1
sth;echocancel=yes
sthdevices=2
sth
sth
sth***
sthhere s my extensions.conf
sth
sth[general]
sthstatic=yes
sthwriteprotect=no
sth
sth[globals]
sthCONSOLE=Console/dsp ; Console interface for demo
sthTRUNK=CAPI
sth
sth[default]
sth
sth; SIP to SIP
sthexten = 100,1,Dial(SIP/joao)
sthexten = 101,1,Dial(SIP/encoder)
sth
sth;SIP to Siemens
sthexten = 118,1,Dial,CAPI/12345678:b${EXTEN}|30
sthexten = 136,1,Dial,CAPI/12345678:b${EXTEN}|30
sthexten = 139,1,Dial,CAPI/12345678:b${EXTEN}|30
sthexten = 116,1,Dial,CAPI/12345678:b${EXTEN}|30
sthexten = 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 
sthuser joao
sth
sthexten = 555,1,Dial(SIP/joao) ; ok, this is the one that doesnt work, 
sthhow can I route the calls?
sth
sth
sth___
sthAsterisk-Users mailing list
sthAsterisk-Users@lists.digium.com
sthhttp://lists.digium.com/mailman/listinfo/asterisk-users
sthTo UNSUBSCRIBE or update options visit:
sth   http://lists.digium.com/mailman/listinfo/asterisk-users
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[Asterisk-Users] routing in extensions.conf

2005-04-22 Thread Joao Pereira 
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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Re: [Asterisk-Users] routing in extensions.conf

2005-04-22 Thread Stefan Helbing 
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==
sthVon: Joao Pereira [EMAIL PROTECTED]
sthDatum: 2005-04-22 18:25:17
sthAn: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
sthBetreff: [Asterisk-Users] routing in extensions.conf
sth
sthHello all,
sthIm using chan_capi to connect from a Siemens High Path to a Aterisk, 
sthwhen I call from the Asterisk clients to the Siemens PBX, it works, when 
sthI call from a Siemens client to a SIP(Asterisk) client, it doesnt work 
sthand says this:
sth
sth  == Starting CAPI[contr1/930]/1 at default,930,1 failed so falling back 
sthto exten 's'
sth  == Starting CAPI[contr1/930]/1 at default,s,1 still failed so falling 
sthback to context 'default'
sthApr 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
sthI think the problem is in the extensions.conf configuration, when the 
sthSiemens calls the Asterisk, it starts ringing and nothing happens, but 
sthwhat do I have to put in the extensions.conf  to route the calls to the 
sthcorrect SIP user?
sthThanks
sthJoao
sth
sth***
sthhere s my capi.conf
sth
sth[general]
sthnationalprefix=0
sthinternationalprefix=00
sthrxgain=0.8
sthtxgain=0.8
sth
sth[interfaces]
sthmsn=12345678
sthincomingmsn=*
sthcontroller=1
sthsoftdtmf=1
sthaccountcode=
sthcontext=default
sth;echosquelch=1
sth;echocancel=yes
sthdevices=2
sth
sth
sth***
sthhere s my extensions.conf
sth
sth[general]
sthstatic=yes
sthwriteprotect=no
sth
sth[globals]
sthCONSOLE=Console/dsp ; Console interface for demo
sthTRUNK=CAPI
sth
sth[default]
sth
sth; SIP to SIP
sthexten = 100,1,Dial(SIP/joao)
sthexten = 101,1,Dial(SIP/encoder)
sth
sth;SIP to Siemens
sthexten = 118,1,Dial,CAPI/12345678:b${EXTEN}|30
sthexten = 136,1,Dial,CAPI/12345678:b${EXTEN}|30
sthexten = 139,1,Dial,CAPI/12345678:b${EXTEN}|30
sthexten = 116,1,Dial,CAPI/12345678:b${EXTEN}|30
sthexten = 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 
sthuser joao
sth
sthexten = 555,1,Dial(SIP/joao) ; ok, this is the one that doesnt work, 
sthhow can I route the calls?
sth
sth
sth___
sthAsterisk-Users mailing list
sthAsterisk-Users@lists.digium.com
sthhttp://lists.digium.com/mailman/listinfo/asterisk-users
sthTo UNSUBSCRIBE or update options visit:
sth   http://lists.digium.com/mailman/listinfo/asterisk-users
sth
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