For ZAP cards, you can tell Asterisk to answer calls immediately across trunks. Does CAPI have the same type of setting? I am not familiar with Asterisk and CAPI so I am not sure of the options.
In Zapata.conf, setting immediate=yes will make the call drop into the 's' extension of the context. Setting immediate=no is supposed to make Asterisk wait until a valid extension is dialed (I have had little to no success with this portion of the setting). If you can change a similar setting for CAPI, you should be able to drop into a non-variable extension in the context (ie. <> i, s, t, etc.). -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joao Pereira Sent: Wednesday, August 10, 2005 7:59 AM To: Asterisk Users Mailing List - Non-Commercial Discussion; [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] dialplan defenition But to have a transparent integration with VoIP and legacy, I cant make users dial twice... or having to whait for Asterisks dialtone, and dial the number. I whant to dial the 74XXX from a PBX extension (74118 for example) and the IP phone rings. Asterisk just need to forward the 74XXX calls, thats why I think the solution is close to this: exten => _74XXX,1,Dial(SIP/[EMAIL PROTECTED],30,r) ... but it always answers: pbx.c:1877 ast_pbx_run: Channel 'CAPI[contr1/118]/1' sent into invalid extension 's' in context 'default', but no invalid handler Why is CAPI sending it to 's' if I explicitly write "Dial(SIP/[EMAIL PROTECTED],30,r)" ?? João Matt Riddell wrote: > Joao Pereira wrote: > >> Hello list, >> Im writing my dial plan, in witch every SIP phone begins with 74 and >> has more 3 numbers (like 74XXX). >> So, I want to route all 74XXX calls to my sip channel. For this I >> wrote this line: >> exten => s,1,Dial(SIP/[EMAIL PROTECTED],30,r) > > > What is happening is that capi is sending it to s. > > You will need to either set up an IVR, asking which number to send it to. > > So, you would do the following: > > exten => s,1,Answer() > exten => s,2,Background(pls-entr-extn) > exten => _74XXX,1,Dial(SIP/${EXTEN}) > exten => _74XXX,2,Goto(s|1) > exten => _74XXX,102,Goto(s|1) > > You will obviously need to record the pls-entr-extn sound. > > You can do this by making an exten like this: > > exten => 678,1,Record(pls-entr-extn) > _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users