Re: [asterisk-users] ISDN - SIP

2010-07-07 Thread Gergo Csibra 
Friday, June 11, 2010, 12:27:08 AM, Tzafrir wrote:

 On Fri, Jun 11, 2010 at 12:19:37AM +0200, Gergo Csibra wrote:
 Okay. There's some problems with mISDN v2: I'm unable to compile
 zaphfc, because there's no source for it. mISDN v2 works with hfcpci
 too?

 Certainly there is.

 It's also part of the standard dahdi-extra patch. See
 http://git.tzafrir.org.il/?p=dahdi-extra.git;a=tree
 http://svn.debian.org/viewsvn/pkg-voip/dahdi-linux/trunk/debian/patches/dahdi_linux_extra

OK. Last time I checked (2009. dec) there wasn't :)

I downloaded dahdi-extra snapshot, and dahdi from asterisk.org,
untared, I have two directories:

dahdi-extra
dahdi-linux-complete-2.3.0.1+2.3.0

What's next?

I don't understand where to start make with MODULES_EXTRA and
SUBDIRS_EXTRA parameters, and how can I configure drivers...


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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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Re: [asterisk-users] ISDN - SIP

2010-06-11 Thread Philipp von Klitzing 
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: 5 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


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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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Re: [asterisk-users] ISDN - SIP

2010-06-10 Thread Tzafrir Cohen 
On Fri, Jun 11, 2010 at 12:19:37AM +0200, Gergo Csibra wrote:
 Thursday, June 10, 2010, 11:19:16 PM, Philipp wrote:
 
  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).
 
 Okay. There's some problems with mISDN v2: I'm unable to compile
 zaphfc, because there's no source for it. mISDN v2 works with hfcpci
 too?

Certainly there is.

It's also part of the standard dahdi-extra patch. See
http://git.tzafrir.org.il/?p=dahdi-extra.git;a=tree
http://svn.debian.org/viewsvn/pkg-voip/dahdi-linux/trunk/debian/patches/dahdi_linux_extra

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+972-50-7952406   mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir

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Re: [asterisk-users] ISDN - SIP

2010-06-10 Thread Philipp von Klitzing 
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


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Re: [asterisk-users] ISDN - SIP

2010-06-10 Thread Gergo Csibra 
Thursday, June 10, 2010, 11:19:16 PM, Philipp wrote:

 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).

Okay. There's some problems with mISDN v2: I'm unable to compile
zaphfc, because there's no source for it. mISDN v2 works with hfcpci
too?

-- 
Best regards,
 Gergomailto:csi...@gmail.com


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Re: [Asterisk-Users] ISDN/SIP videophone gatewaying?

2005-02-23 Thread Roy Sigurd Karlsbakk 
sure, but what about using asterisk?
On Feb 22, 2005, at 12:39, googleplex wrote:
google for inalp isdn sip gateway
On Tue, 22 Feb 2005 12:23:39 +0100, Roy Sigurd Karlsbakk
[EMAIL PROTECTED] wrote:
Hi
Is it, or could it be possible to gateway from ISDN videophones to IP
videophoning with asterisk using libpri/zaptel etc?
roy
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RE: [Asterisk-Users] ISDN/SIP videophone gatewaying?

2005-02-23 Thread Florian Overkamp 
Hi Roy, 

 -Original Message-
 sure, but what about using asterisk?
 
 On Feb 22, 2005, at 12:39, googleplex wrote:
 
  google for inalp isdn sip gateway

Asterisk currently doesn't understand the ISDN video side. If you use one of
those gateways you probably could get it to interop with SIP videophones via
Asterisk.

Florian 


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[Asterisk-Users] ISDN/SIP videophone gatewaying?

2005-02-22 Thread Roy Sigurd Karlsbakk 
Hi
Is it, or could it be possible to gateway from ISDN videophones to IP 
videophoning with asterisk using libpri/zaptel etc?

roy
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Re: [Asterisk-Users] ISDN/SIP videophone gatewaying?

2005-02-22 Thread googleplex 
google for inalp isdn sip gateway 


On Tue, 22 Feb 2005 12:23:39 +0100, Roy Sigurd Karlsbakk
[EMAIL PROTECTED] wrote:
 Hi
 
 Is it, or could it be possible to gateway from ISDN videophones to IP
 videophoning with asterisk using libpri/zaptel etc?
 
 roy
 
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