Re: [asterisk-users] ISDN - SIP

2010-06-11 Thread 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 - 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=5,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: 5 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: 5 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: 5 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-0007 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=0xdead

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.

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[asterisk-users] ISDN - SIP

2010-06-10 Thread Stefan Dreyer 
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.

My extension conf is:

general]
static=yes
writeprotect=no

[globals]
OUT_PORT=1

[ISDN]
exten = 12345,1,Dial(SIP/012346737...@sipprovider.local)


If i call to the msn 12345, the SIP-call is going out, but after a
second the call is stopped.
What is wrong, with my configuration?

Kernel show
Jun 10 20:48:58 wolf kernel: hdlc_down unknown prim(280)
Jun 10 20:49:04 wolf kernel: MDL_ERROR|REQ (tei_l2)

Asterisk shows:


P[ 1] MGMT: SSTATUS: L1_ACTIVATED

P[ 1] handle_frm: frm-addr:42000103 frm-prim:3f082

P[ 1] channel with stid:0 not in use!

P[ 1] handle_frm: frm-addr:42000103 frm-prim:30582

P[ 1] set_channel: bc-channel:0 channel:1

P[ 1] I IND :NEW_CHANNEL oad:xxx dad:12345 pid:2 state:none

P[ 1]  -- channel:1 mode:TE cause:16 ocause:16 rad: cad:

P[ 1]  -- info_dad: onumplan:2 dnumplan:4 rnumplan:  cpnnumplan:0

P[ 1]  -- caps:Speech pi:0 keypad: sending_complete:1

P[ 1]  -- screen:0 -- pres:0

P[ 1]  -- addr:0 l3id:20007 b_stid:0 layer_id:0

P[ 1]  -- facility:Fac_None out_facility:Fac_None

P[ 1]  -- bc_state:BCHAN_CLEANED

P[ 1] Chan not existing at the moment bc-l3id:20007 bc:0x8721e9c
event:NEW_CHANNEL port:1 channel:1
P[ 1] NO USERUESRINFO

P[ 1]  -- found chan (preselected): 1

P[ 1] set_chan_in_stack: 1

P[ 1] setup_bc: with dsp

P[ 1]  -- Channel is 1

P[ 1]  -- TRANSPARENT Mode

P[ 1] I IND :SETUP oad:xxx dad:12345 pid:2 state:none

P[ 1]  -- channel:1 mode:TE cause:16 ocause:16 rad: cad:

P[ 1]  -- info_dad: onumplan:2 dnumplan:4 rnumplan:  cpnnumplan:0

P[ 1]  -- caps:Speech pi:0 keypad: sending_complete:1

P[ 1]  -- screen:0 -- pres:0

P[ 1]  -- addr:50010102 l3id:20007 b_stid:10010100 layer_id:50010180

P[ 1]  -- facility:Fac_None out_facility:Fac_None

P[ 1]  -- bc_state:BCHAN_ACTIVATED

P[ 1]  -- Bearer: Speech

P[ 1]  -- Codec: Alaw

P[ 0]  -- * NEW CHANNEL dad:12345 oad:xxx

P[ 1] read_config: Getting Config

P[ 1]  -- CTON: Unknown

P[ 1]  -- EXPORT_PID: pid:2

P[ 1]  -- PRES: Allowed (0)

P[ 1]  -- SCREEN: Unscreened (0)

P[ 1] * Queuing chan 0x89e5410

P[ 1] I SEND:RELEASE oad:xxx dad:12345 pid:2

P[ 1]  -- bc_state:BCHAN_ACTIVATED

P[ 1]  -- channel:1 mode:TE cause:16 ocause:1 rad: cad:

P[ 1]  -- info_dad: onumplan:2 dnumplan:4 rnumplan:  cpnnumplan:0

P[ 1]  -- caps:Speech pi:0 keypad: sending_complete:1

P[ 1]  -- screen:0 -- pres:0

P[ 1]  -- addr:50010102 l3id:20007 b_stid:10010100 layer_id:50010180

P[ 1]  -- facility:Fac_None out_facility:Fac_None

P[ 1] GOT SETUP OK

P[ 1] Sending msg, prim:34d80 addr:41000104 dinfo:20007

P[ 1] BCHAN: bchan ACT Confirm pid:2

P[ 1] handle_frm: frm-addr:42000103 frm-prim:3f182

P[ 1]  -- lib: RELEASE_CR Ind with l3id:20007
P[ 1]  -- lib: CLEANING UP l3id: 20007
P[ 1]  -- hangup
P[ 1] * IND : HANGUPpid:2 ctx:ISDN dad:12345 oad: State:EXTCANTMATCH
P[ 1]  -- l3id:20007
P[ 1]  -- cause:16
P[ 1]  -- out_cause:16
P[ 1]  -- Channel: mISDN/1-u0 hungup new state:CLEANING
P[ 1] $$$ CLEANUP CALLED pid:2
P[ 1] $$$ Cleaning up bc with stid :10010100 pid:2
P[ 1]  -- ec_disable
P[ 1] Sending Control ECHOCAN_OFF
P[ 1] ph_control: c1:2319 c2:0
P[ 1] empty_chan_in_stack: 1
P[ 0] handle_bchan: BC not found for prim:f2481 with addr:55010180 dinfo:0
P[ 0] received 1k Unhandled Bchannel Messages: prim f2481 len 0 from
addr 55010180, dinfo 0 on this port.
P[ 1] MGMT: SSTATUS: L1_DEACTIVATED



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