Asterisk & Lancom1722

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achilles87

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Hallo zusammen,

ich sitze jetzt schon seit 2 Tagen an folgendem Problem:

Ich habe hier ein Gerät von Lancom das als Media Gateway zwischen SIP und ISDN dienen soll. Ich würde ihn gerne mit meinem Asterisk Server "verbinden". Ich finde aber keine funktionierende Lösung.

sip.conf:
Code: 
[general]
register => 1000:1000@and-lancom1722
context=fax-in
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes

[lancom]
type=friend
username=1000
fromuser=1000
secret=1000
host=and-lancom1722
fromdomain=and-lancom1722
qualify=yes
insecure=port,invite

extensions.conf:
Code: 
[default]
exten => _X.,1,Dial(SIP/${EXTEN}@lancom)

[sonstige]

[meine-telefone]
exten => _X.,1,Dial(SIP/${EXTEN}@lancom)

[fax-in]
exten => _X.,1,Dial(IAX2/iaxmodem)

Habe im Lancom den User 1000 angelegt und wenn ich mit einem Softphone, das auch am Lancom mit einer anderen Nummer registriert ist, die 1000 anrufe, will der Lancom den Anruf auch auf den User "1000", der ja Asterisk ist, weiterleiten. Leider wird der Anruf aber als abgelehnt bezeichnet und die CLI spuckt auch keinerlei Meldungen aus...

Was mache ich falsch???

Mit freundlichen Grüßen
achilles87
 
Zuletzt bearbeitet:
Noch ein Nachtrag:

Der SIP-User 1000 von Lancom wird von Asterisk erkannt und registriert
Code: 
and-fax*CLI> sip show registry
Host                            Username       Refresh State                Reg.Time
and-lancom1722:5060             1000               105 Registered           Wed, 30 Sep 2009 15:00:14

Wenn ich den User 1001(ist angemeldet (x-lite) und auch im lancom eingerichtet) anrufe passiert folgendes...

Code: 
and-fax*CLI> console dial 1001
[Sep 30 15:03:42] WARNING[12276]: chan_oss.c:686 setformat: Unable to re-open DSP device /dev/dsp: No such file or directory
    -- Executing [1001@default:1] Dial("Console/dsp", "SIP/1001@lancom") in new stack
    -- Called 1001@lancom
    -- SIP/lancom-083f9738 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
  == Auto fallthrough, channel 'Console/dsp' status is 'CONGESTION'
 << Hangup on console >>

Code: 
and-fax*CLI> sip show peers
Name/username              Host            Dyn Nat ACL Port     Status
lancom/1000                10.1.3.29                   5060     OK (5 ms)
1 sip peers [Monitored: 1 online, 0 offline Unmonitored: 0 online, 0 offline]

Code: 
and-fax*CLI> sip show peer lancom
and-fax*CLI>

  * Name       : lancom
  Secret       : <Set>
  MD5Secret    : <Not set>
  Context      : fax-in
  Subscr.Cont. : <Not set>
  Language     :
  AMA flags    : Unknown
  Transfer mode: open
  CallingPres  : Presentation Allowed, Not Screened
  FromUser     : 1000
  FromDomain   : and-lancom1722
  Callgroup    :
  Pickupgroup  :
  Mailbox      :
  VM Extension : asterisk
  LastMsgsSent : 32767/65535
  Call limit   : 0
  Dynamic      : No
  Callerid     : "" <>
  MaxCallBR    : 384 kbps
  Expire       : -1
  Insecure     : port,invite
  Nat          : RFC3581
  ACL          : No
  T38 pt UDPTL : No
  CanReinvite  : Yes
  PromiscRedir : No
  User=Phone   : No
  Video Support: No
  Trust RPID   : No
  Send RPID    : No
  Subscriptions: Yes
  Overlap dial : Yes
  DTMFmode     : rfc2833
  LastMsg      : 0
  ToHost       : and-lancom1722
  Addr->IP     : 10.1.3.29 Port 5060
  Defaddr->IP  : 0.0.0.0 Port 0
  Def. Username: 1000
  SIP Options  : (none)
  Codecs       : 0x8000e (gsm|ulaw|alaw|h263)
  Codec Order  : (none)
  Auto-Framing:  No
  Status       : OK (5 ms)
  Useragent    :
  Reg. Contact :

SIP DEBUG sagt:
Code: 
and-fax*CLI> console dial 1001
[Sep 30 15:36:11] WARNING[17073]: chan_oss.c:686 setformat: Unable to re-open DSP device /dev/dsp: No such file or directory
    -- Executing [1001@default:1] Dial("Console/dsp", "SIP/1001@lancom") in new stack
Audio is at 10.1.1.209 port 19718
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.1.3.29:5060:
INVITE sip:1001@and-lancom1722 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.209:5060;branch=z9hG4bK2d03b5a1;rport
From: "asterisk" <sip:1000@and-lancom1722>;tag=as54d50338
To: <sip:1001@and-lancom1722>
Contact: <sip:[email protected]>
Call-ID: 3d737693632a56622793e6773ea8073a@and-lancom1722
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 30 Sep 2009 13:36:11 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Type: application/sdp
Content-Length: 283

v=0
o=root 12244 12244 IN IP4 10.1.1.209
s=session
c=IN IP4 10.1.1.209
t=0 0
m=audio 19718 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
    -- Called 1001@lancom
and-fax*CLI>
<--- SIP read from 10.1.3.29:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 10.1.1.209:5060;branch=z9hG4bK2d03b5a1;rport
From: "asterisk"<sip:1000@and-lancom1722>;tag=as54d50338
To: <sip:1001@and-lancom1722>
Call-ID: 3d737693632a56622793e6773ea8073a@and-lancom1722
CSeq: 102 INVITE
Max-Forwards: 70
User-Agent: LANCOM 1722 VoIP (Annex B) / 7.60.0160 / 25.02.2009
Server: and-lancom1722
Allow: REGISTER, INVITE, ACK, CANCEL, BYE, REFER, NOTIFY, OPTIONS
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---
Transmitting (no NAT) to 10.1.3.29:5060:
ACK sip:1001@and-lancom1722 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.209:5060;branch=z9hG4bK2d03b5a1;rport
From: "asterisk" <sip:1000@and-lancom1722>;tag=as54d50338
To: <sip:1001@and-lancom1722>
Contact: <sip:[email protected]>
Call-ID: 3d737693632a56622793e6773ea8073a@and-lancom1722
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
    -- SIP/lancom-083f4618 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
  == Auto fallthrough, channel 'Console/dsp' status is 'CONGESTION'
Really destroying SIP dialog '3d737693632a56622793e6773ea8073a@and-lancom1722' Method: INVITE
 << Hangup on console >>
Reliably Transmitting (no NAT) to 10.1.3.29:5060:
OPTIONS sip:and-lancom1722 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.209:5060;branch=z9hG4bK5e88dceb;rport
From: "asterisk" <sip:[email protected]>;tag=as12c0be0a
To: <sip:and-lancom1722>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 30 Sep 2009 13:36:23 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0


---
and-fax*CLI>
<--- SIP read from 10.1.3.29:5060 --->
SIP/2.0 481 Call/Transaction Does Not Exist
Via: SIP/2.0/UDP 10.1.1.209:5060;branch=z9hG4bK5e88dceb;rport
From: "asterisk"<sip:[email protected]>;tag=as12c0be0a
To: <sip:and-lancom1722>
Call-ID: [email protected]
CSeq: 102 OPTIONS
Max-Forwards: 70
User-Agent: LANCOM 1722 VoIP (Annex B) / 7.60.0160 / 25.02.2009
Server: and-lancom1722
Allow: REGISTER, INVITE, ACK, CANCEL, BYE, REFER, NOTIFY, OPTIONS
Supported: replaces
Content-Length: 0


<------------->
--- (12 headers 0 lines) ---
Really destroying SIP dialog '[email protected]' Method: OPTIONS
Really destroying SIP dialog '[email protected]' Method: REGISTER
Reliably Transmitting (no NAT) to 10.1.3.29:5060:
OPTIONS sip:and-lancom1722 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.209:5060;branch=z9hG4bK04c7a3d3;rport
From: "asterisk" <sip:[email protected]>;tag=as27221a20
To: <sip:and-lancom1722>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 30 Sep 2009 13:37:23 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0


---
and-fax*CLI>
<--- SIP read from 10.1.3.29:5060 --->
SIP/2.0 481 Call/Transaction Does Not Exist
Via: SIP/2.0/UDP 10.1.1.209:5060;branch=z9hG4bK04c7a3d3;rport
From: "asterisk"<sip:[email protected]>;tag=as27221a20
To: <sip:and-lancom1722>
Call-ID: [email protected]
CSeq: 102 OPTIONS
Max-Forwards: 70
User-Agent: LANCOM 1722 VoIP (Annex B) / 7.60.0160 / 25.02.2009
Server: and-lancom1722
Allow: REGISTER, INVITE, ACK, CANCEL, BYE, REFER, NOTIFY, OPTIONS
Supported: replaces
Content-Length: 0


<------------->
--- (12 headers 0 lines) ---
Really destroying SIP dialog '[email protected]' Method: OPTIONS
and-fax*CLI> sip debug off
Usage: sip set debug
       Enables dumping of SIP packets for debugging purposes

       sip set debug ip <host[:PORT]>
       Enables dumping of SIP packets to and from host.

       sip set debug peer <peername>
       Enables dumping of SIP packets to and from host.
       Require peer to be registered.
The 'sip debug' command is deprecated and will be removed in a future release. Please use 'sip set debug' instead.
[Sep 30 15:37:38] NOTICE[12255]: chan_sip.c:7753 sip_reregister:    -- Re-registration for  1000@and-lancom1722
REGISTER 12 headers, 0 lines
Reliably Transmitting (no NAT) to 10.1.3.29:5060:
REGISTER sip:and-lancom1722 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.209:5060;branch=z9hG4bK636a69f6;rport
From: <sip:1000@and-lancom1722>;tag=as11ca5489
To: <sip:1000@and-lancom1722>
Call-ID: [email protected]
CSeq: 105 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Expires: 120
Contact: <sip:[email protected]>
Event: registration
Content-Length: 0


---
and-fax*CLI>
<--- SIP read from 10.1.3.29:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.1.209:5060;branch=z9hG4bK636a69f6;rport
From: <sip:1000@and-lancom1722;user=phone>;tag=as11ca5489
To: <sip:1000@and-lancom1722>;tag=-401203375-832928408
Call-ID: [email protected]
CSeq: 105 REGISTER
Max-Forwards: 70
Server: and-lancom1722
Allow: REGISTER, INVITE, ACK, CANCEL, BYE, REFER, NOTIFY, OPTIONS
Contact: <sip:[email protected]:5060>;expires=120
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---
Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: REGISTER)
[Sep 30 15:37:38] NOTICE[12255]: chan_sip.c:13064 handle_response_register: Outbound Registration: Expiry for and-lancom1722 is 120 sec (Scheduling reregistration in 105 s)
Really destroying SIP dialog '[email protected]' Method: REGISTER
Reliably Transmitting (no NAT) to 10.1.3.29:5060:
OPTIONS sip:and-lancom1722 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.209:5060;branch=z9hG4bK4a559031;rport
From: "asterisk" <sip:[email protected]>;tag=as79937514
To: <sip:and-lancom1722>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 30 Sep 2009 13:38:23 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0


---
and-fax*CLI>
<--- SIP read from 10.1.3.29:5060 --->
SIP/2.0 481 Call/Transaction Does Not Exist
Via: SIP/2.0/UDP 10.1.1.209:5060;branch=z9hG4bK4a559031;rport
From: "asterisk"<sip:[email protected]>;tag=as79937514
To: <sip:and-lancom1722>
Call-ID: [email protected]
CSeq: 102 OPTIONS
Max-Forwards: 70
User-Agent: LANCOM 1722 VoIP (Annex B) / 7.60.0160 / 25.02.2009
Server: and-lancom1722
Allow: REGISTER, INVITE, ACK, CANCEL, BYE, REFER, NOTIFY, OPTIONS
Supported: replaces
Content-Length: 0


<------------->
--- (12 headers 0 lines) ---
Really destroying SIP dialog '[email protected]' Method: OPTIONS

Und hier noch der SIP trace vom Lancom bei einem Anruf vom Softphone 1001 zu 1000 (Asterisk):
Code: 
[SIP-Packet] 2009/09/30 16:58:35,940 [PACKET] :
Receiving datagram with length 770 from 10.1.3.128:3016 to 10.1.3.29:5060
INVITE sip:[email protected] SIP/2.0\r\n
Via: SIP/2.0/UDP 10.1.3.128:3016;branch=z9hG4bK-d8754z-343aff1e1b7ea46a-1---d875
4z-;rport\r\n
Max-Forwards: 70\r\n
Contact: <sip:[email protected]:3016>\r\n
To: "1000"<sip:[email protected]>\r\n
From: "1001"<sip:[email protected]>;tag=300cbc35\r\n
Call-ID: MTkyMzEwMzZjMTRhMjYzNjJmYTY0OGNjNmViOTIxNjY.\r\n
CSeq: 1 INVITE\r\n
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INF
O\r\n
Content-Type: application/sdp\r\n
User-Agent: X-Lite release 1103k stamp 53621\r\n
Content-Length: 256\r\n
\r\n
v=0\r\n
o=- 2 2 IN IP4 10.1.3.128\r\n
s=CounterPath X-Lite 3.0\r\n
c=IN IP4 10.1.3.128\r\n
t=0 0\r\n
m=audio 3472 RTP/AVP 107 0 8 101\r\n
a=alt:1 1 : qETcts5d V99s++vi 10.1.3.128 3472\r\n
a=fmtp:101 0-15\r\n
a=rtpmap:107 BV32/16000\r\n
a=rtpmap:101 telephone-event/8000\r\n
a=sendrecv\r\n

[SIP-Packet] 2009/09/30 16:58:35,940 [PACKET] :
Sending datagram with length 456 from 10.1.3.29:5060 to 10.1.3.128:3016
SIP/2.0 100 Trying\r\n
Via: SIP/2.0/UDP 10.1.3.128:3016;branch=z9hG4bK-d8754z-343aff1e1b7ea46a-1---d875
4z-;rport\r\n
From: "1001"<sip:[email protected]>;tag=300cbc35\r\n
To: "1000"<sip:[email protected]>\r\n
Call-ID: MTkyMzEwMzZjMTRhMjYzNjJmYTY0OGNjNmViOTIxNjY.\r\n
CSeq: 1 INVITE\r\n
Max-Forwards: 70\r\n
User-Agent: LANCOM 1722 VoIP (Annex B) / 7.60.0160 / 25.02.2009\r\n
Server: and-lancom1722\r\n
Allow: REGISTER, INVITE, ACK, CANCEL, BYE, REFER, NOTIFY, OPTIONS\r\n
Content-Length: 0\r\n
\r\n

[SIP-Packet] 2009/09/30 16:58:35,980 [PACKET] :
Sending datagram with length 436 from 10.1.3.29:5060 to 10.1.3.128:3016
SIP/2.0 488 Not Acceptable Here\r\n
Via: SIP/2.0/UDP 10.1.3.128:3016;branch=z9hG4bK-d8754z-343aff1e1b7ea46a-1---d875
4z-;rport\r\n
From: "1001"<sip:[email protected];user=phone>;tag=300cbc35\r\n
To: <sip:[email protected]>;tag=1180587207--1619019769\r\n
Call-ID: MTkyMzEwMzZjMTRhMjYzNjJmYTY0OGNjNmViOTIxNjY.\r\n
CSeq: 1 INVITE\r\n
Max-Forwards: 70\r\n
Server: and-lancom1722\r\n
Allow: REGISTER, INVITE, ACK, CANCEL, BYE, REFER, NOTIFY, OPTIONS\r\n
Content-Length: 0\r\n
\r\n

[SIP-Packet] 2009/09/30 16:58:35,980 [PACKET] :
Receiving datagram with length 312 from 10.1.3.128:3016 to 10.1.3.29:5060
ACK sip:[email protected] SIP/2.0\r\n
Via: SIP/2.0/UDP 10.1.3.128:3016;branch=z9hG4bK-d8754z-343aff1e1b7ea46a-1---d875
4z-;rport\r\n
To: <sip:[email protected]>;tag=1180587207--1619019769\r\n
From: "1001"<sip:[email protected]>;tag=300cbc35\r\n
Call-ID: MTkyMzEwMzZjMTRhMjYzNjJmYTY0OGNjNmViOTIxNjY.\r\n
CSeq: 1 ACK\r\n
Content-Length: 0\r\n
\r\n

HILFEEEEEEEEEEEEE!!!!!
 
Zuletzt bearbeitet:
Kann mir niemand sagen, wie ich ein Lancom-Gerät mit einem Asterisk-Server verbinden kann???

Ich brauche HILFE!!!
 
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