Asterisk INVITE error

[yangp]

Hallo !

Ich bitte fuer die hilfe in Deutsche oder Englische sprache.

Hier liegt das problem:

Ich moechte die nummer 400 anrufen, welcher an Asterisk-1 server registriert ist.
Ich rufe aus den nummer 2970, welcher an Asterisk-2 registriert ist.

Edit Guard-X: Bitte nächstes mal Code-Tags verwenden!

Asterisk-1 config file:
sip.conf

[btctrunk]
type=peer
username=
secret=
nat=yes
disallow=all
allow=ulaw
allow=alaw
allow=g729
qualify=yes
context=uplink
host=212.103.134.179

extensions.conf

[from-local-users]
; hinti
exten => 300,hint,SIP/300
exten => 400,hint,SIP/400
exten => 600,hint,SIP/600
exten => 700,hint,SIP/700
exten => 800,hint,SIP/800
exten => 802,hint,SIP/802

exten => 400,1,Dial(SIP/400)
exten => 400,2,VoiceMail(u400)
exten => 400,3,Playback(goodbye)
exten => 400,4,Hangup
exten => 400,102,VoiceMail(u400)
exten => 400,103,Playback(goodbye)
exten => 400,104,Hangup

Asterisk-2 config file:

sip.conf

#
[asterisk-jan]
#
type=peer
#
username=
#
secret=
#
nat=yes
#
disallow=all
#
allow=ulaw
#
allow=alaw
#
allow=g729
#
qualify=yes
#
context=btc
#
host=212.103.133.6
#
 
#

extensions.conf

#
exten => _X.,2,Hangup
#
exten => _9.,1,mcc2(${EXTEN:1}|direct|IAX2/btctr1/${EXTEN:1})
#
exten => _9.,2,Hangup
#
exten => _29XX,1,Dial(SIP/${EXTEN},60,tr)
#
exten => _29XX,2,Hangup
#
exten => _6.,1,Dial(SIP/pritrunk/${EXTEN:1})
#
exten => _6.,2,Hangup
#
exten => _400,1,Dial(SIP/${EXTEN}@asterisk-jan,60,t)
#
exten => _400,2,Hangup

Error code
Das logging ist aus dem asterisk-1 server (account 400).


error:

#
-- Executing Dial("SIP/2970-0ba14fe8", "SIP/400@asterisk-jan|60|t") in new stack
#
-- Called 400@asterisk-jan
#
-- Got SIP response 486 "Busy Here" back from 212.103.133.6
#
-- SIP/asterisk-jan-0b9d7fa0 is busy

und noch...

 -- Executing Set("SIP/212.103.134.179-b7958750", "MONITOR_FILENAME=20071128-135825-400-"2970" <2970>-out") in new stack
    -- Executing Monitor("SIP/212.103.134.179-b7958750", "wav|20071128-135825-400-"2970" <2970>-out|mb") in new stack
    -- Executing Dial("SIP/212.103.134.179-b7958750", "SIP/400@btctrunk|60|t") in new stack
    -- Called 400@btctrunk
Nov 28 13:58:25 NOTICE[21869]: chan_sip.c:9750 handle_response_invite: Failed to authenticate on INVITE to '"2970" <sip:[email protected]>;tag=as19b535fd'
    -- SIP/btctrunk-08154150 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
    -- Executing Busy("SIP/212.103.134.179-b7958750", "6") in new stack
  == Spawn extension (uplink, 400, 4) exited non-zero on 'SIP/212.103.134.179-b7958750'
Nov 28 13:58:25 ERROR[21869]: chan_sip.c:11408 sipsock_read: We could NOT get the channel lock for SIP/212.103.134.179-b7958750! 
Nov 28 13:58:25 ERROR[21869]: chan_sip.c:11409 sipsock_read: SIP MESSAGE JUST IGNORED: ACK 
Nov 28 13:58:25 ERROR[21869]: chan_sip.c:11410 sipsock_read: BAD! BAD! BAD!

OK Nochmal die debug lines:

v=0
o=root 21851 21851 IN IP4 212.103.133.6
s=session
c=IN IP4 212.103.133.6
t=0 0
m=audio 18470 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---

<-- SIP read from 212.103.134.179:5060: 
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 212.103.133.6:5060;branch=z9hG4bK74e04bfd;received=212.103.133.6;rport=5060
From: "2970" <sip:[email protected]>;tag=as29496e42
To: <sip:[email protected]>;tag=as4a6a6d73
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="47e86fe0"
Content-Length: 0


--- (10 headers 0 lines) ---
Transmitting (NAT) to 212.103.134.179:5060:
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 212.103.133.6:5060;branch=z9hG4bK74e04bfd;rport
From: "2970" <sip:[email protected]>;tag=as29496e42
To: <sip:[email protected]>;tag=as4a6a6d73
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 ACK
User-Agent: asterisk
Max-Forwards: 70
Content-Length: 0

---
Nov 28 14:33:47 NOTICE[21869]: chan_sip.c:9750 handle_response_invite: Failed to authenticate on INVITE to '"2970" <sip:[email protected]>;tag=as29496e42'                                                                       
    -- SIP/btctrunk-08154150 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
    -- Executing Busy("SIP/212.103.134.179-b7958750", "6") in new stack
Transmitting (NAT) to 212.103.134.179:5060:
SIP/2.0 486 Busy Here
Via: SIP/2.0/UDP 212.103.134.179:5060;branch=z9hG4bK18ce527c;received=212.103.134.179;rport=5060
From: "2970" <sip:[email protected]>;tag=as73649e04
To: <sip:[email protected]>;tag=as0c1c2c3c
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing


---
  == Spawn extension (uplink, 400, 4) exited non-zero on 'SIP/212.103.134.179-b7958750'

<-- SIP read from 212.103.134.179:5060: 
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 212.103.134.179:5060;branch=z9hG4bK18ce527c;rport
From: "2970" <sip:[email protected]>;tag=as73649e04
To: <sip:[email protected]>;tag=as0c1c2c3c
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

--- (10 headers 0 lines) ---
Nov 28 14:33:47 ERROR[21869]: chan_sip.c:11408 sipsock_read: We could NOT get the channel lock for SIP/212.103.134.179-b7958750!                                                                                                  
Nov 28 14:33:47 ERROR[21869]: chan_sip.c:11409 sipsock_read: SIP MESSAGE JUST IGNORED: ACK 
Nov 28 14:33:47 ERROR[21869]: chan_sip.c:11410 sipsock_read: BAD! BAD! BAD!
Destroying call '[email protected]'
Destroying call '[email protected]'

Vielen dank /Thank you !

 

[cheGGo]

Hi,

du musst auf beiden Asterisk Servern ein

insecure=very

in die peer config eintrage.

zum beispiel:

[btctrunk]
type=peer
[B]insecure=very[/B]
username=
secret=
nat=yes
disallow=all
allow=ulaw
allow=alaw
allow=g729
qualify=yes
context=uplink
host=212.103.134.179

gruß cheGGo