Hi! > But if i try to establish ISDN->SIP-Dialout, the redirection ist not > working.
Your logs are very sketchy and difficult to understand because you stripped them of some details and cut out lines in between. > From: "55555" <sip:s...@sip>;tag=as1ec770c5 This line does not make much sense. > exten => 123456,1,Dial(SIP/987...@sip) > exten => 123457,1,Dial(SIP/33) > ; both not working. Do i need to accept the call before? What is the CLI output of: "sip show peer sip" and "sip show peer 33"? Note: It it not good practice to define local sip peers (phones) with numbers only (like 33). Use alphanumeric names like "phone1" or "mac11223344566". > The Call is rejected whith the message "No Connection" (de: "kein > Anschluss unter dieser Nummer"). ... > -- chan_capi queue frame:[ TYPE: Control (4) SUBCLASS: Hangup (1) ] Yes, that is what you get: A hangup cause code of "1", which means "number not allocated". Use the dialplan variables ${HANGUPCAUSE} and ${DIALSTATUS} to process accordingly this in extensions.conf. So: Obviously you dialed the wrong number. ;-> > INVITE sip:987...@sip SIP/2.0 > To: <sip:987...@sip> > What is wrong. An why SIP-to internal SIP-Phone(/33) See above "sip show peer 33". Maybe you haven't registered the phone, or you have forgotten to give it a static IP in sip.conf. Philipp -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users