Asterisk 1.6 und Sipgate - ich krieg es nicht hin

BirdyB · 26 Feb 2010

Hallo,
ich habe auf einem vServer Asterisk installiert und alle via SIP verbundenen Telefone können auch problemlos miteinander telefonieren.
Jetzt habe ich meinen Sipgate-Account gemäß der verschiedenen Anleitungen über das Webinterface in Asterisk eingebunden. In der Übersicht erscheint der Account mit Status 105.
Bei den Incoming Calling Rules habe ich erstmal eine Catchall-Regel erstellt. Wenn ich jetzt meine Sipgate-Nummer anwähle, dann erscheint auch einiges an Text in der Console aber unter anderem immer "IGNORING THIS INVITE".

Welche Einstellungen muss ich noch überprüfen? Woran könnte es liegen, dass das nicht funktioniert? Ich bin ziemlich ratlos!

kombjuder · 26 Feb 2010

BirdyB schrieb:
H

Welche Einstellungen muss ich noch überprüfen? Woran könnte es liegen, dass das nicht funktioniert? Ich bin ziemlich ratlos!

Da ich meine Glaskugel verlegt habe, brauche ich mal die dir angezeigte Ausgabe bei einem eingehenden Anruf und die Asterisk Version.
Am besten auch den Ausschnitt aus der extensions.conf.

Nachdem eine Ausgabe kommt, scheint die sip.conf richtig zu sein.

BirdyB · 27 Feb 2010

So, hier die Ausgabe vom Log:

Code:

<------------->
--- (12 headers 0 lines) ---
Really destroying SIP dialog '[email protected]' Method: OPTIONS
v36302*CLI> 
<--- SIP read from UDP:217.10.79.9:5060 --->
INVITE sip:[email protected] SIP/2.0
Record-Route: <sip:217.10.79.9;lr=on;ftag=as791923e4>
Record-Route: <sip:172.20.40.1;lr=on>
Record-Route: <sip:217.10.79.9;lr=on;ftag=as791923e4>
Via: SIP/2.0/UDP 217.10.79.9:5060;branch=z9hG4bK0757.cb3a3f27.1
Via: SIP/2.0/UDP 172.20.40.1;branch=z9hG4bK0757.cb3a3f27.1
Via: SIP/2.0/UDP 217.10.79.9:5060;received=217.10.68.222;branch=z9hG4bK55534d18
Via: SIP/2.0/UDP 217.116.117.69:5060;branch=z9hG4bK55534d18;rport=5060
From: "01773838516" <sip:[email protected]>;tag=as791923e4
To: <sip:[email protected]>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 INVITE
Max-Forwards: 67
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 412

v=0
o=root 4243 4243 IN IP4 217.116.117.69
s=session
c=IN IP4 217.116.117.69
t=0 0
m=audio 17406 RTP/AVP 8 0 3 97 18 112 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:112 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------->
--- (18 headers 19 lines) ---
Ignoring this INVITE request
       > HTTP Manager add header action: ping
v36302*CLI> 
<--- SIP read from UDP:217.10.79.9:5060 --->
INVITE sip:[email protected] SIP/2.0
Record-Route: <sip:217.10.79.9;lr=on;ftag=as791923e4>
Record-Route: <sip:172.20.40.1;lr=on>
Record-Route: <sip:217.10.79.9;lr=on;ftag=as791923e4>
Via: SIP/2.0/UDP 217.10.79.9:5060;branch=z9hG4bK0757.cb3a3f27.1
Via: SIP/2.0/UDP 172.20.40.1;branch=z9hG4bK0757.cb3a3f27.1
Via: SIP/2.0/UDP 217.10.79.9:5060;received=217.10.68.222;branch=z9hG4bK55534d18
Via: SIP/2.0/UDP 217.116.117.69:5060;branch=z9hG4bK55534d18;rport=5060
From: "01773838516" <sip:[email protected]>;tag=as791923e4
To: <sip:[email protected]>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 INVITE
Max-Forwards: 67
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 412

v=0
o=root 4243 4243 IN IP4 217.116.117.69
s=session
c=IN IP4 217.116.117.69
t=0 0
m=audio 17406 RTP/AVP 8 0 3 97 18 112 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:112 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------->
--- (18 headers 19 lines) ---
Ignoring this INVITE request
       > HTTP Manager add header action: ping
v36302*CLI> 
<--- SIP read from UDP:217.10.79.9:5060 --->
INVITE sip:[email protected] SIP/2.0
Record-Route: <sip:217.10.79.9;lr=on;ftag=as791923e4>
Record-Route: <sip:172.20.40.1;lr=on>
Record-Route: <sip:217.10.79.9;lr=on;ftag=as791923e4>
Via: SIP/2.0/UDP 217.10.79.9:5060;branch=z9hG4bK0757.cb3a3f27.1
Via: SIP/2.0/UDP 172.20.40.1;branch=z9hG4bK0757.cb3a3f27.1
Via: SIP/2.0/UDP 217.10.79.9:5060;received=217.10.68.222;branch=z9hG4bK55534d18
Via: SIP/2.0/UDP 217.116.117.69:5060;branch=z9hG4bK55534d18;rport=5060
From: "01773838516" <sip:[email protected]>;tag=as791923e4
To: <sip:[email protected]>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 INVITE
Max-Forwards: 67
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 412

v=0
o=root 4243 4243 IN IP4 217.116.117.69
s=session
c=IN IP4 217.116.117.69
t=0 0
m=audio 17406 RTP/AVP 8 0 3 97 18 112 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:112 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------->
--- (18 headers 19 lines) ---
Ignoring this INVITE request
Really destroying SIP dialog '[email protected]' Method: REGISTER
Really destroying SIP dialog '[email protected]' Method: REGISTER
       > HTTP Manager add header action: ping
v36302*CLI> 
<--- SIP read from UDP:217.10.79.9:5060 --->
INVITE sip:[email protected] SIP/2.0
Record-Route: <sip:217.10.79.9;lr=on;ftag=as791923e4>
Record-Route: <sip:172.20.40.1;lr=on>
Record-Route: <sip:217.10.79.9;lr=on;ftag=as791923e4>
Via: SIP/2.0/UDP 217.10.79.9:5060;branch=z9hG4bK0757.cb3a3f27.1
Via: SIP/2.0/UDP 172.20.40.1;branch=z9hG4bK0757.cb3a3f27.1
Via: SIP/2.0/UDP 217.10.79.9:5060;received=217.10.68.222;branch=z9hG4bK55534d18
Via: SIP/2.0/UDP 217.116.117.69:5060;branch=z9hG4bK55534d18;rport=5060
From: "01773838516" <sip:[email protected]>;tag=as791923e4
To: <sip:[email protected]>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 INVITE
Max-Forwards: 67
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 412

v=0
o=root 4243 4243 IN IP4 217.116.117.69
s=session
c=IN IP4 217.116.117.69
t=0 0
m=audio 17406 RTP/AVP 8 0 3 97 18 112 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:112 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------->
--- (18 headers 19 lines) ---
Ignoring this INVITE request

Und der Dialplan

Code:

v36302*CLI> dialplan show
[ Context 'app_dial_gosub_virtual_context' created by 'app_dial' ]
  's' =>            1. NoOp()                                     [app_dial]
v36302*CLI> 
[ Context 'app_queue_gosub_virtual_context' created by 'app_queue' ]
  's' =>            1. NoOp()                                     [app_queue]
v36302*CLI> 
[ Context 'parkedcalls' created by 'features' ]
  '700' =>          1. Park()                                     [features]
v36302*CLI> 
[ Context 'DLPN_AllAccess' created by 'pbx_config' ]
  Include =>        'CallingRule_outgoing'                        [pbx_config]
  Include =>        'default'                                     [pbx_config]
v36302*CLI> 
[ Context 'DID_trunk_1_timeinterval_everytime' created by 'pbx_config' ]
  's' =>            1. Goto(default,6003,1)                       [pbx_config]
v36302*CLI> 
[ Context 'CallingRule_outgoing' created by 'pbx_config' ]
  's' =>            1. Macro(trunkdial-failover-0.3,${trunk_1}/${EXTEN:0},,trunk_1,) [pbx_config]
v36302*CLI> 
[ Context 'DID_trunk_1_default' created by 'pbx_config' ]
v36302*CLI> 
[ Context 'DID_trunk_1' created by 'pbx_config' ]
  Include =>        'DID_trunk_1_timeinterval_everytime,*,mon-sun,*,*' [pbx_config]
  Include =>        'DID_trunk_1_default'                         [pbx_config]
v36302*CLI> 
[ Context 'macro-trunkdial-failover-0.3' created by 'pbx_config' ]
  '1-CHANUNAVAIL' => 1. Dial(${ARG2})                              [pbx_config]
                    2. Hangup()                                   [pbx_config]
  '1-CONGESTION' => 1. Dial(${ARG2})                              [pbx_config]
                    2. Hangup()                                   [pbx_config]
  '1-dial' =>       1. Dial(${ARG1})                              [pbx_config]
                    2. Gotoif(${LEN(${ARG2})} > 0 ?1-${DIALSTATUS},1:1-out,1) [pbx_config]
  '1-fmsetcid' =>   1. Set(CALLERID(num)=${FMCIDNUM})             [pbx_config]
                    2. Set(CALLERID(name)=${FMCIDNAME})           [pbx_config]
                    3. Goto(1-dial,1)                             [pbx_config]
  '1-out' =>        1. Hangup()                                   [pbx_config]
  '1-setgbobname' => 1. Set(CALLERID(name)=${GLOBAL_OUTBOUNDCIDNAME}) [pbx_config]
                    2. Goto(s,3)                                  [pbx_config]
  's' =>            1. GotoIf($[${LEN(${FMCIDNUM})} > 6]?1-fmsetcid,1) [pbx_config]
                    2. GotoIf($[${LEN(${GLOBAL_OUTBOUNDCIDNAME})} > 1]?1-setgbobname,1) [pbx_config]
                    3. Set(CALLERID(num)=${IF($[${LEN(${CID_${CALLERID(num)}})} > 2]?${CID_${CALLERID(num)}}:)}) [pbx_config]
                    4. GotoIf($[${LEN(${CALLERID(num)})} > 6]?1-dial,1) [pbx_config]
                    5. Set(CALLERID(all)=${IF($[${LEN(${CID_${ARG3}})} > 6]?${CID_${ARG3}}:${GLOBAL_OUTBOUNDCID})}) [pbx_config]
                    6. Goto(1-dial,1)                             [pbx_config]
v36302*CLI> 
[ Context 'asterisk_guitools' created by 'pbx_config' ]
  'executecommand' => 1. System(${command})                         [pbx_config]
                    2. Hangup()                                   [pbx_config]
  'play_file' =>    1. Answer()                                   [pbx_config]
                    2. Playback(${var1})                          [pbx_config]
                    3. Hangup()                                   [pbx_config]
  'record_vmenu' => 1. Answer()                                   [pbx_config]
                    2. Playback(vm-intro)                         [pbx_config]
                    3. Record(${var1},0,500,k)                    [pbx_config]
                    4. Playback(vm-saved)                         [pbx_config]
                    5. Playback(vm-goodbye)                       [pbx_config]
                    6. Hangup()                                   [pbx_config]
v36302*CLI> 
[ Context 'pagegroups' created by 'pbx_config' ]
v36302*CLI> 
[ Context 'page_an_extension' created by 'pbx_config' ]
v36302*CLI> 
[ Context 'directory' created by 'pbx_config' ]
  '6666' =>         1. Directory(default,default,)                [pbx_config]
v36302*CLI> 
[ Context 'voicemailgroups' created by 'pbx_config' ]
v36302*CLI> 
[ Context 'voicemenus' created by 'pbx_config' ]
v36302*CLI> 
[ Context 'queues' created by 'pbx_config' ]
v36302*CLI> 
[ Context 'ringgroups' created by 'pbx_config' ]
v36302*CLI> 
[ Context 'conferences' created by 'pbx_config' ]
  '6300' =>         1. MeetMe(${EXTEN},)                          [pbx_config]
v36302*CLI> 
[ Context 'macro-pagingintercom' created by 'pbx_config' ]
  's' =>            1. SIPAddHeader(Alert-Info: ${PAGING_HEADER}) [pbx_config]
                    2. Page(${ARG1},${ARG2})                      [pbx_config]
                    3. Hangup()                                   [pbx_config]
v36302*CLI> 
[ Context 'macro-stdexten-followme' created by 'pbx_config' ]
  'a' =>            1. VoicemailMain(${ORIG_ARG1})                [pbx_config]
  's' =>            1. Answer()                                   [pbx_config]
                    2. Set(ORIG_ARG1=${ARG1})                     [pbx_config]
                    3. Dial(${ARG2},${RINGTIME},${DIALOPTIONS})   [pbx_config]
                    4. Set(__FMCIDNUM=${CALLERID(num)})           [pbx_config]
                    5. Set(__FMCIDNAME=${CALLERID(name)})         [pbx_config]
                    6. Followme(${ORIG_ARG1},${FOLLOWMEOPTIONS})  [pbx_config]
                    7. Voicemail(${ORIG_ARG1},u)                  [pbx_config]
  's-BUSY' =>       1. Voicemail(${ORIG_ARG1},b)                  [pbx_config]
                    2. Goto(default,s,1)                          [pbx_config]
  's-NOANSWER' =>   1. Voicemail(${ORIG_ARG1},u)                  [pbx_config]
  '_s-.' =>         1. Goto(s-NOANSWER,1)                         [pbx_config]
v36302*CLI> 
[ Context 'macro-stdexten' created by 'pbx_config' ]
  'a' =>            1. VoicemailMain(${ORIG_ARG1})                [pbx_config]
  's' =>            1. Set(__DYNAMIC_FEATURES=${FEATURES})        [pbx_config]
                    2. Set(ORIG_ARG1=${ARG1})                     [pbx_config]
                    3. GotoIf($["${FOLLOWME_${ARG1}}" = "1"]?6:4) [pbx_config]
                    4. Dial(${ARG2},${RINGTIME},${DIALOPTIONS})   [pbx_config]
                    5. Goto(s-${DIALSTATUS},1)                    [pbx_config]
                    6. Macro(stdexten-followme,${ARG1},${ARG2})   [pbx_config]
  's-BUSY' =>       1. Voicemail(${ORIG_ARG1},b)                  [pbx_config]
                    2. Goto(default,s,1)                          [pbx_config]
  's-NOANSWER' =>   1. Voicemail(${ORIG_ARG1},u)                  [pbx_config]
                    2. Goto(default,s,1)                          [pbx_config]
  '_s-.' =>         1. Goto(s-NOANSWER,1)                         [pbx_config]
v36302*CLI> 
[ Context 'default' created by 'pbx_config' ]
  '6000' =>         hint: SIP/6000                                [pbx_config]
                    1. Dial(${HINT})                              [pbx_config]
  '6001' =>         hint: SIP/6001                                [pbx_config]
                    1. Dial(${HINT})                              [pbx_config]
  '6002' =>         hint: SIP/6002                                [pbx_config]
                    1. Macro(stdexten,6002,${HINT})               [pbx_config]
  '6003' =>         hint: SIP/6003                                [pbx_config]
                    1. Macro(stdexten,6003,${HINT})               [pbx_config]
  '6004' =>         hint: SIP/6004                                [pbx_config]
                    1. Dial(${HINT})                              [pbx_config]
  '6999' =>         1. VoiceMailMain(${CALLERID(num)}@default)    [pbx_config]
  'a' =>            1. VoicemailMain(${MBOX})                     [pbx_config]
  '_#6XXX' =>       1. Set(MBOX=${EXTEN:1}@default)               [pbx_config]
                    2. VoiceMail(${MBOX})                         [pbx_config]
v36302*CLI> 
[ Context 'page' created by 'pbx_config' ]
  '_X.' =>          1. Macro(page,SIP/${EXTEN})                   [pbx_config]
v36302*CLI> 
[ Context 'macro-page' created by 'pbx_config' ]
  '76245' =>        1. Macro(page,SIP/Grandstream1)               [pbx_config]
  's' =>            1. ChanIsAvail(${ARG1},s)                     [pbx_config]
                    2. GoToIf([${AVAILSTATUS} = "1"]?autoanswer:fail) [pbx_config]
     [autoanswer]   3. Set(_ALERT_INFO="RA")                      [pbx_config]
                    4. SIPAddHeader(Call-Info: Answer-After=0)    [pbx_config]
                    5. NoOp()                                     [pbx_config]
                    6. Dial(${ARG1})                              [pbx_config]
     [fail]         7. Hangup()                                   [pbx_config]
  '_7XXX' =>        1. Macro(page,SIP/${EXTEN})                   [pbx_config]
v36302*CLI> 
[ Context 'macro-trunkdial' created by 'pbx_config' ]
  's' =>            1. Dial(${ARG1})                              [pbx_config]
                    2. Goto(s-${DIALSTATUS},1)                    [pbx_config]
  's-BUSY' =>       1. Hangup()                                   [pbx_config]
  's-NOANSWER' =>   1. Hangup()                                   [pbx_config]
  '_s-.' =>         1. NoOp()                                     [pbx_config]
v36302*CLI> 
-= 44 extensions (92 priorities) in 25 contexts. =-

kombjuder · 27 Feb 2010

BirdyB schrieb:
So, hier die Ausgabe vom Log:

Ist so für mich nicht lesbar.

Was ist das Trix-Box, oder ein anderer Dialplangenerator?

Da deine Fehlermeldung entgegen meinen Vermutungen nicht auf dem cli sondern im debug kommt, wird die sip.conf gebraucht und schreib jetzt endlich, um was für eine Asteriskversion es sich handelt.

BirdyB · 27 Feb 2010

Also Asterisk äußert sich zur Version wie folgt: Asterisk SVN-trunk-r248861
Ich nutze einfach nur das Digium Web-Interface, damit hab ich eigentlich die ganze Konfiguration gemacht
so, hier die sip.conf

Code:

[general]
port = 5060
bindaddr = 0.0.0.0
context = default
qualify=no
disallow=all
allow=alaw
allow=ulaw
allow=g729
allow=gsm
allow=slinear
srvlookup=yes
register => 12345678:[email protected]/12345678

[sipgate-out]
type=friend
insecure=invite
nat=yes
username=12345678
fromuser=12345678
fromdomain=sipgate.de
secret=XXXXX
host=sipgate.de
qualify=yes
canreinvite=no
dtmfmode=rfc2833

kombjuder · 27 Feb 2010

BirdyB schrieb:
Ich nutze einfach nur das Digium Web-Interface, damit hab ich eigentlich die ganze

Dein Context sipgate hat keinen Eintrag context=...
Damit wird der Kontext default benutzt. In default gibt es aber, so wie das aussieht, werder einen Eintrag _X. noch 12345678 sondern nur 6000, 6001 ...

Damit kann das Gespräch nicht angenommen werden. Die Fehlermeldung von Asterisk passt allerdings nicht ganz zu meinem Ergebnis.

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Asterisk 1.6 und Sipgate - ich krieg es nicht hin

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