On 06/10/10 23:19, Philipp von Klitzing wrote: > Hi! > >> i am using asterisk 1.6.2 with mISDN and a passive AVM Fritz!card on >> CentOS 5.5. The only thing, i want to do is a call-redirection from an >> isdn-call to my mobile via sip-account. > > Unless you are using mISDN v2: Do yourself a favour and switch to CAPI > with chan_capi and fcpci. mISDN v1 is a guarantee for headaches (and > unstable systems).
After a little torture to install fcpci, SIP->ISDN-Dialout is working. But if i try to establish ISDN->SIP-Dialout, the redirection ist not working. [isdn-in] ; MSN 123456 -> 987...@sip exten => 123456,1,Dial(SIP/987...@sip) exten => 123457,1,Dial(SIP/33) ; both not working. Do i need to accept the call before? [misdnOut] ; DIAL-Out-Working exten => _0X.,1,Dial(CAPI/contr1/${EXTEN}) [default] include => misdnOut The Call is rejected whith the message "No Connection" (de: "kein Anschluss unter dieser Nummer"). But the outgoing SIP-Call is made. The log shows: -- CONNECT_IND (PLCI=0x101,DID=12345,CID=55555,CIP=0x10,CONTROLLER=0x1) == Started pbx on channel CAPI/ISDN1#02/12345-10 -- Executing [12...@isdn-in:1] Dial("CAPI/ISDN1#02/12345-10", "SIP/87...@sip,45,t") in new stack == Using SIP RTP CoS mark 5 Audio is at 212.x.y.z port 15256 Adding codec 0x8 (alaw) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x4 (ulaw) to SDP Reliably Transmitting (no NAT) to a.b.c.d:5060: INVITE sip:987...@sip SIP/2.0 Via: SIP/2.0/UDP 212.x.y.z:5060;branch=z9hG4bK5a55a76e;rport Max-Forwards: 70 From: "55555" <sip:s...@sip>;tag=as1ec770c5 To: <sip:987...@sip> Contact: <sip:dry...@212.68.91.194> Call-ID: 1979cd9a3c3cb9013e9cd9660cd33...@sip CSeq: 102 INVITE ... Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer ... v=0 o=root 1971852647 1971852647 IN IP4 212.x.y.z s=Asterisk PBX 1.6.2.8 c=IN IP4 212.x.y.z t=0 0 m=audio 15256 RTP/AVP 8 3 0 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called 987...@sip <--- SIP read from UDP:a.b.c.d:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 212.x.y.z:5060;branch=z9hG4bK5a55a76e;rport From: "55555" <sip:s...@sip>;tag=as1ec770c5 To: <sip:98...@sip> Contact: sip:987...@a.b.c.d:5060 Call-ID: 1979cd9a3c3cb9013e9cd9660cd33...@sip CSeq: 102 INVITE Server: (Very nice Sip Registrar/Proxy Server) Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE WWW-Authenticate: Digest realm="sip...",nonce="3042653437",algorithm=MD5 Content-Length: 0 ... --- Audio is at 212.x.y.z port 15256 Adding codec 0x8 (alaw) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x4 (ulaw) to SDP Reliably Transmitting (no NAT) to a.d.c.d:5060: INVITE sip:987...@sip SIP/2.0 Via: SIP/2.0/UDP 212.x.y.z:5060;branch=z9hG4bK51f5e20e;rport Max-Forwards: 70 From: "55555" <sip:s...@sip>;tag=as1ec770c5 To: <sip:987...@sip> Contact: <sip:dry...@212.x.y.z> Call-ID: 1979cd9a3c3cb9013e9cd9660cd33...@sip.voipdiscount.com CSeq: 103 INVITE ... Found RTP audio format 8 Found audio description format PCMA for ID 8 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port a.b.c.d:41302 -- SIP/sip-00000007 is making progress passing it to CAPI/ISDN1#02/12345-10 -- chan_capi queue frame:[ TYPE: Control (4) SUBCLASS: Hangup (1) ] [ISDN1#02] Scheduling destruction of SIP dialog '19....@sip' in 32000 ms (Method: INVITE) Reliably Transmitting (no NAT) to 77.72.169.134:5060: Scheduling destruction of SIP dialog '1...@sip' in 32000 ms (Method: INVITE) == Spawn extension (isdn-in, 12345, 1) exited non-zero on 'CAPI/ISDN1#02/12345-10' == ISDN1#02: Interface cleanup PLCI=0xdead0000 What is wrong. An why SIP-to internal SIP-Phone(/33) is not working. >From internal SIP to ISDN and internal SIP to external SIP is working. -- _____________________________________________________________________ -- 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