asterisk users - Jun 2010 - ISDN -> SIP

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/012346737222 at 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 : HANGUP    pid:2 ctx:ISDN dad:12345 oad:xxxx 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

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).

Philipp

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:sip at sip>;tag=as1ec770c5

This line does not make much sense.
> 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?
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:987654 at sip SIP/2.0
> To: <sip:987654 at 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