[asterisk-users] ISDN -> SIP

[Stefan Dreyer]

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 -> 987654 at Sip
exten => 123456,1,Dial(SIP/987654 at 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 [12345 at isdn-in:1] Dial("CAPI/ISDN1#02/12345-10",
"SIP/87654 at 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:987654 at sip SIP/2.0
Via: SIP/2.0/UDP 212.x.y.z:5060;branch=z9hG4bK5a55a76e;rport

Max-Forwards: 70
From: "55555" <sip:sip at sip>;tag=as1ec770c5

To: <sip:987654 at sip>
Contact: <sip:dryman at 212.68.91.194>
Call-ID: 1979cd9a3c3cb9013e9cd9660cd3331f at 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 987654 at 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:sip at sip>;tag=as1ec770c5
To: <sip:98765 at sip>
Contact: sip:987654 at a.b.c.d:5060
Call-ID: 1979cd9a3c3cb9013e9cd9660cd3331f at 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:987654 at sip SIP/2.0
Via: SIP/2.0/UDP 212.x.y.z:5060;branch=z9hG4bK51f5e20e;rport
Max-Forwards: 70
From: "55555" <sip:sip at sip>;tag=as1ec770c5
To: <sip:987654 at sip>
Contact: <sip:dryman at 212.x.y.z>
Call-ID: 1979cd9a3c3cb9013e9cd9660cd3331f at 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...f at sip' in 32000 ms (Method:
INVITE)
Reliably Transmitting (no NAT) to 77.72.169.134:5060:

Scheduling destruction of SIP dialog '19f at 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.