[Problem] Mal wieder Telekom ... Asterisk 13.22 + FreePBX 13.0 (IncrediblePBX 13-13-7)

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bofh42

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Moin,

da mir meine "alte" Asterisk-Anlage leider gestorben ist (Hardware), musste ich eine neue aufsetzen:

Hardware: Raspberry Pi 3+
OS: Debian jessie
Asterisk 13.22
FreePBX 13.0
gebundelt von Ward Mundy Inc (IncrediblePBX 13-13-7) (Avantfax etc etc)

Aber erst mal geht es mal wieder um die Telekom (und ggf noch SIPGate + PersonalVoIP)

Das ganze läuft mit chan_sip (pjsip ging gar nichts).
Die Anlage steht - genau wie die Vorgängeranlege) im internen Netz hinter eine IPFire Firewall, die wiederum über eine FritzBOX am VDSL der Telekom hängt

Offenbar ist in der Konfig ganz grundsätzlich der Wurm drin:

SIP-Anruf kommt über Telekom rein:
SNGREP sagt


Code: 
INVITE sip:[email protected]:5060 SIP/2.0
             217.0.27.53:5060             192.168.80.12:5060            217.0.20.192:1065 │Max-Forwards: 50
          ──────────┬─────────          ──────────┬─────────          ──────────┬─────────│Via: SIP/2.0/UDP 217.0.27.53:5060;branch=z9hG4bKg3Zqkv7i851bq9k6ibqejzdbojsfifqtn
  15:24:44.813389   │        INVITE (SDP)         │                             │         │To: <sip:[email protected];user=phone>
        +0.002581   │ ──────────────────────────> │                             │         │From: <sip:[email protected];user=phone>;tag=h7g4Esbg_p65543t1546957484m748540c298543244s1_459049648-325714831
  15:24:44.815970   │                             │        INVITE (SDP)         │         │Call-ID: p65543t1546957484m748540c298543244s2
        +0.498772   │                             │ <────────────────────────── │         │CSeq: 1 INVITE
  15:24:45.314742   │        INVITE (SDP)         │                             │         │Contact: <sip:[email protected];transport=udp>;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel"
        +0.002101   │ ────────────────────────>>> │                             │         │Record-Route: <sip:217.0.27.53;transport=udp;lr>
  15:24:45.316843   │                             │        INVITE (SDP)         │         │Accept-Contact: *;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel"
        +0.998850   │                             │ <<<──────────────────────── │         │Min-Se: 900
  15:24:46.315693   │        INVITE (SDP)         │                             │         │P-Asserted-Identity: <sip:[email protected];user=phone>
        +0.002484   │ ────────────────────────>>> │                             │         │Session-Expires: 1800
  15:24:46.318177   │                             │        INVITE (SDP)         │         │Supported: timer
        +1.998373   │                             │ <<<──────────────────────── │         │Supported: 100rel
  15:24:48.316550   │        INVITE (SDP)         │                             │         │Supported: histinfo
        +0.003444   │ ────────────────────────>>> │                             │         │Content-Type: application/sdp
  15:24:48.319994   │                             │        INVITE (SDP)         │         │Content-Length: 172
        +3.997360   │                             │ <<<──────────────────────── │         │Session-ID: dfd24f0efff9340437f7b9fc71ce6cd4
  15:24:52.317354   │        INVITE (SDP)         │                             │         │Allow: REGISTER, REFER, NOTIFY, SUBSCRIBE, PRACK, UPDATE, PUBLISH, INFO, INVITE, ACK, OPTIONS, CANCEL, BYE
        +0.003406   │ ────────────────────────>>> │                             │         │Accept: application/vnd.etsi.sci+xml
  15:24:52.320760   │                             │        INVITE (SDP)         │         │Accept: multipart/mixed
        +7.997695   │                             │ <<<──────────────────────── │         │Accept: application/vnd.telekom.service_indication+xml
  15:25:00.318455   │        INVITE (SDP)         │                             │         │Accept: application/vnd.etsi.cug+xml
        +0.002982   │ ────────────────────────>>> │                             │         │Accept: application/sdp
  15:25:00.321437   │                             │        INVITE (SDP)         │         │
       +15.998477   │                             │ <<<──────────────────────── │         │v=0
  15:25:16.319914   │        INVITE (SDP)         │                             │         │o=- 18098394 459049437 IN IP4 217.0.4.197
        +0.002989   │ ────────────────────────>>> │                             │         │s=Basic Session
  15:25:16.322903   │                             │        INVITE (SDP)         │         │c=IN IP4 217.0.4.197
       +62.300015   │                             │ <<<──────────────────────── │         │t=0 0
  15:26:18.622918   │      401 Unauthorized       │                             │         │m=audio 60664 RTP/AVP 8 99
                    │ <────────────────────────── │                             │         │a=rtpmap:99 telephone-event/8000
                    │                             │                             │         │a=fmtp:99 0-15


bzw als zweites

Code: 
INVITE sip:[email protected]:5060 SIP/2.0
             217.0.27.53:5060             192.168.80.12:5060            217.0.20.192:1065 │Max-Forwards: 50
          ──────────┬─────────          ──────────┬─────────          ──────────┬─────────│Via: SIP/2.0/UDP 217.0.20.192:1065;branch=z9hG4bKg3Zqkv7ij9hqico3pqlwem8v7cce9h7ik
  15:24:44.813389   │        INVITE (SDP)         │                             │         │To: <sip:[email protected];user=phone>
        +0.002581   │ ──────────────────────────> │                             │         │From: <sip:[email protected];user=phone>;tag=h7g4Esbg_p65543t1546957484m748540c298543244s1_459049648-325714831
  15:24:44.815970   │                             │        INVITE (SDP)         │         │Call-ID: p65543t1546957484m748540c298543244s2
        +0.498772   │                             │ <────────────────────────── │         │CSeq: 1 INVITE
  15:24:45.314742   │        INVITE (SDP)         │                             │         │Contact: <sip:[email protected]:1065;transport=udp>;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel"
        +0.002101   │ ────────────────────────>>> │                             │         │Record-Route: <sip:217.0.20.192;transport=udp;lr>
  15:24:45.316843   │                             │        INVITE (SDP)         │         │Accept-Contact: *;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel"
        +0.998850   │                             │ <<<──────────────────────── │         │Min-Se: 900
  15:24:46.315693   │        INVITE (SDP)         │                             │         │P-Asserted-Identity: <sip:[email protected];user=phone>
        +0.002484   │ ────────────────────────>>> │                             │         │Session-Expires: 1800
  15:24:46.318177   │                             │        INVITE (SDP)         │         │Supported: timer
        +1.998373   │                             │ <<<──────────────────────── │         │Supported: 100rel
  15:24:48.316550   │        INVITE (SDP)         │                             │         │Supported: histinfo
        +0.003444   │ ────────────────────────>>> │                             │         │Content-Type: application/sdp
  15:24:48.319994   │                             │        INVITE (SDP)         │         │Content-Length: 172
        +3.997360   │                             │ <<<──────────────────────── │         │Session-ID: dfd24f0efff9340437f7b9fc71ce6cd4
  15:24:52.317354   │        INVITE (SDP)         │                             │         │Allow: REGISTER, REFER, NOTIFY, SUBSCRIBE, PRACK, UPDATE, PUBLISH, INFO, INVITE, ACK, OPTIONS, CANCEL, BYE
        +0.003406   │ ────────────────────────>>> │                             │         │Accept: application/vnd.etsi.sci+xml
  15:24:52.320760   │                             │        INVITE (SDP)         │         │Accept: multipart/mixed
        +7.997695   │                             │ <<<──────────────────────── │         │Accept: application/vnd.telekom.service_indication+xml
  15:25:00.318455   │        INVITE (SDP)         │                             │         │Accept: application/vnd.etsi.cug+xml
        +0.002982   │ ────────────────────────>>> │                             │         │Accept: application/sdp
  15:25:00.321437   │                             │        INVITE (SDP)         │         │
       +15.998477   │                             │ <<<──────────────────────── │         │v=0
  15:25:16.319914   │        INVITE (SDP)         │                             │         │o=- 18098394 459049437 IN IP4 217.0.5.148
        +0.002989   │ ────────────────────────>>> │                             │         │s=Basic Session
  15:25:16.322903   │                             │        INVITE (SDP)         │         │c=IN IP4 217.0.5.148
       +62.300015   │                             │ <<<──────────────────────── │         │t=0 0
  15:26:18.622918   │      401 Unauthorized       │                             │         │m=audio 52324 RTP/AVP 8 99
                    │ <────────────────────────── │                             │         │a=rtpmap:99 telephone-event/8000
                    │                             │                             │         │a=fmtp:99 0-15
                    │                             │                             │         │
                    │                             │                             │         │


und ganz zum Schluß:
Code: 
SIP/2.0 401 Unauthorized
             217.0.27.53:5060             192.168.80.12:5060            217.0.20.192:1065 │Via: SIP/2.0/UDP 217.0.27.53:5060;branch=z9hG4bKg3Zqkv7i851bq9k6ibqejzdbojsfifqtn;received=217.0.27.53;rport=5060
          ──────────┬─────────          ──────────┬─────────          ──────────┬─────────│From: <sip:[email protected];user=phone>;tag=h7g4Esbg_p65543t1546957484m748540c298543244s1_459049648-325714831
  15:24:44.813389   │        INVITE (SDP)         │                             │         │To: <sip:[email protected];user=phone>;tag=as32ef163f
        +0.002581   │ ──────────────────────────> │                             │         │Call-ID: p65543t1546957484m748540c298543244s2
  15:24:44.815970   │                             │        INVITE (SDP)         │         │CSeq: 1 INVITE
        +0.498772   │                             │ <────────────────────────── │         │Server: FPBX-13.0.192.19(13.22.0)
  15:24:45.314742   │        INVITE (SDP)         │                             │         │Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
        +0.002101   │ ────────────────────────>>> │                             │         │Supported: replaces, timer
  15:24:45.316843   │                             │        INVITE (SDP)         │         │WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="52c35b27"
        +0.998850   │                             │ <<<──────────────────────── │         │Content-Length: 0


In der Asterisk-Console (und auch im log-File) kommt davon NICHTS, kein einziges Zeichen an ...


sip show peers sagt aber:

Code: 
incrediblepbx2*CLI> sip show peers
Name/username             Host                                    Dyn Forcerport Comedia    ACL Port     Status      Description
6000/6000                 192.168.80.161                           D  Yes        Yes         A  5060     OK (20 ms)
6001/6001                 192.168.80.161                           D  Yes        Yes         A  5060     OK (18 ms)
6002/6002                 192.168.80.162                           D  Yes        Yes         A  5060     OK (32 ms)
6003/6003                 192.168.80.165                           D  Yes        Yes         A  5060     OK (12 ms)
6004/6004                 192.168.80.8                             D  Yes        Yes         A  52308    OK (11 ms)
6005/6005                 192.168.80.164                           D  Yes        Yes         A  5060     OK (13 ms)
6006                      (Unspecified)                            D  Yes        Yes         A  0        UNKNOWN
6007/6007                 192.168.80.162                           D  Yes        Yes         A  5066     OK (69 ms)
6008                      (Unspecified)                            D  Yes        Yes         A  0        UNKNOWN
6009/6009                 (Unspecified)                            D  Yes        Yes         A  0        UNKNOWN
701                       (Unspecified)                            D  Yes        Yes         A  0        UNKNOWN
BlueSIP/bluesip/christian 217.74.179.29                               Yes        Yes            5060     OK (20 ms)
PersonalVoIP-1-chan_sip/5 46.182.250.46                               Yes        Yes            5060     OK (13 ms)
PersonalVoIP-2-chan_sip/5 46.182.250.46                               Yes        Yes            5060     OK (13 ms)
PersonalVoIP-3-chan_sip/5 46.182.250.46                               Yes        Yes            5060     OK (12 ms)
SipGate-chan_sip/2294444e 217.10.79.9                                 Yes        Yes            5060     OK (13 ms)
telekom-924xxxy-chansip/0 217.0.20.192                                Yes        Yes            5060     OK (23 ms)
telekom-924xxxz-chansip/0 217.0.20.192                                Yes        Yes            5060     OK (24 ms)
telekom-924xxxa-chansip/0 217.0.20.192                                Yes        Yes            5060     OK (27 ms)
telekom-924xxxb-chansip/0 217.0.20.192                                Yes        Yes            5060     OK (75 ms)
telekom-924xxxc-sip/05173 217.0.20.192                                Yes        Yes            5060     OK (23 ms)
telekom-925xxd-chansip/ch 217.0.20.192                                Yes        Yes            5060     OK (21 ms)
telekom-925xxe-chansip/05 217.0.20.192                                Yes        Yes            5060     OK (1018 ms)
telekom-925xxf-chansip/05 217.0.20.192                                Yes        Yes            5060     OK (1019 ms)
24 sip peers [Monitored: 20 online, 4 offline Unmonitored: 0 online, 0 offline]
    -- Registered SIP '6009' at 10.80.0.6:51616
incrediblepbx2*CLI>


Dann wählen wir doch mal raus ....


Auf der console: NICHTS. Gar nichts. Auch nicht im Log

aber in sngrep:


Die Trunk-Settings sehen so aus:

Code: 
username=<E-MAIL ohne @T- :::>
defaultuser=<E-MAIL ohne @T- :::>
type=peer
secret=<password>
remotepassword=<password>
qualify=yes
nat=yes
insecure=very
host=tel.t-online.de
fromuser=telefonnummermitvorwahl
fromdomain=tel.t-online.de
dtmfmode=rfc2833
canreinvite=no
directmedia=yes

register: telefonnummer-mit-vorwahl:password:[email protected]~481

--------------------------------------------------------

OK, das war nix.

Nächster Versuch:

Code: 
type=friend
username=telefonnummer-mit-vorwahl
fromuser=telefonnummer-mit-vorwahl
secret=password
host=tel.t-online.de
nat=yes
dtmfmode=rfc2833
canreinvite=update
fromdomain=tel.t-online.de
insecure=very
qualify=yes

diesmal gibt es was in der console:

Code: 
incrediblepbx2*CLI>
  == Using SIP VIDEO TOS bits 136
  == Using SIP VIDEO CoS mark 6
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
       > 0x6ce82b80 -- Strict RTP learning after remote address set to: 10.80.0.6:32674
       > 0x6cec2be8 -- Strict RTP learning after remote address set to: 10.80.0.6:32676
    -- Executing [0531zzzz@from-internal:1] Macro("SIP/6009-0000001a", "user-callerid,LIMIT,EXTERNAL,") in new stack
    -- Executing [s@macro-user-callerid:1] Set("SIP/6009-0000001a", "TOUCH_MONITOR=1546958788.26") in new stack
    -- Executing [s@macro-user-callerid:2] Set("SIP/6009-0000001a", "AMPUSER=6009") in new stack
    -- Executing [s@macro-user-callerid:3] GotoIf("SIP/6009-0000001a", "0?report") in new stack
    -- Executing [s@macro-user-callerid:4] ExecIf("SIP/6009-0000001a", "1?Set(REALCALLERIDNUM=6009)") in new stack
    -- Executing [s@macro-user-callerid:5] Set("SIP/6009-0000001a", "AMPUSER=6009") in new stack
    -- Executing [s@macro-user-callerid:6] GotoIf("SIP/6009-0000001a", "0?limit") in new stack
    -- Executing [s@macro-user-callerid:7] Set("SIP/6009-0000001a", "AMPUSERCIDNAME=Name Mobil2") in new stack
    -- Executing [s@macro-user-callerid:8] GotoIf("SIP/6009-0000001a", "0?report") in new stack
    -- Executing [s@macro-user-callerid:9] Set("SIP/6009-0000001a", "AMPUSERCID=6009") in new stack
    -- Executing [s@macro-user-callerid:10] Set("SIP/6009-0000001a", "__DIAL_OPTIONS=Ttr") in new stack
    -- Executing [s@macro-user-callerid:11] Set("SIP/6009-0000001a", "CALLERID(all)="Name Mobil2" <6009>") in new stack
    -- Executing [s@macro-user-callerid:12] GotoIf("SIP/6009-0000001a", "0?limit") in new stack
    -- Executing [s@macro-user-callerid:13] ExecIf("SIP/6009-0000001a", "1?Set(GROUP(concurrency_limit)=6009)") in new stack
    -- Executing [s@macro-user-callerid:14] GotoIf("SIP/6009-0000001a", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,27)
    -- Executing [s@macro-user-callerid:27] Set("SIP/6009-0000001a", "CALLERID(number)=6009") in new stack
    -- Executing [s@macro-user-callerid:28] Set("SIP/6009-0000001a", "CALLERID(name)=Name Mobil2") in new stack
    -- Executing [s@macro-user-callerid:29] GotoIf("SIP/6009-0000001a", "0?cnum") in new stack
    -- Executing [s@macro-user-callerid:30] Set("SIP/6009-0000001a", "CDR(cnam)=Name Mobil2") in new stack
    -- Executing [s@macro-user-callerid:31] Set("SIP/6009-0000001a", "CDR(cnum)=6009") in new stack
    -- Executing [s@macro-user-callerid:32] Set("SIP/6009-0000001a", "CHANNEL(language)=de") in new stack
    -- Executing [0531zzzzz@from-internal:2] Gosub("SIP/6009-0000001a", "sub-record-check,s,1(out,0531zzzzzz,dontcare)") in new stack
    -- Executing [s@sub-record-check:1] GotoIf("SIP/6009-0000001a", "0?initialized") in new stack
    -- Executing [s@sub-record-check:2] Set("SIP/6009-0000001a", "__REC_STATUS=INITIALIZED") in new stack
    -- Executing [s@sub-record-check:3] Set("SIP/6009-0000001a", "NOW=1546958788") in new stack
    -- Executing [s@sub-record-check:4] Set("SIP/6009-0000001a", "__DAY=08") in new stack
    -- Executing [s@sub-record-check:5] Set("SIP/6009-0000001a", "__MONTH=01") in new stack
    -- Executing [s@sub-record-check:6] Set("SIP/6009-0000001a", "__YEAR=2019") in new stack
    -- Executing [s@sub-record-check:7] Set("SIP/6009-0000001a", "__TIMESTR=20190108-154628") in new stack
    -- Executing [s@sub-record-check:8] Set("SIP/6009-0000001a", "__FROMEXTEN=6009") in new stack
    -- Executing [s@sub-record-check:9] Set("SIP/6009-0000001a", "__MON_FMT=wav") in new stack
    -- Executing [s@sub-record-check:10] NoOp("SIP/6009-0000001a", "Recordings initialized") in new stack
    -- Executing [s@sub-record-check:11] ExecIf("SIP/6009-0000001a", "0?Set(ARG3=dontcare)") in new stack
    -- Executing [s@sub-record-check:12] Set("SIP/6009-0000001a", "REC_POLICY_MODE_SAVE=") in new stack
    -- Executing [s@sub-record-check:13] ExecIf("SIP/6009-0000001a", "0?Set(REC_STATUS=NO)") in new stack
    -- Executing [s@sub-record-check:14] GotoIf("SIP/6009-0000001a", "3?checkaction") in new stack
    -- Goto (sub-record-check,s,17)
    -- Executing [s@sub-record-check:17] GotoIf("SIP/6009-0000001a", "1?sub-record-check,out,1") in new stack
    -- Goto (sub-record-check,out,1)
    -- Executing [out@sub-record-check:1] NoOp("SIP/6009-0000001a", "Outbound Recording Check from 6009 to 0531zzzzz") in new stack
    -- Executing [out@sub-record-check:2] Set("SIP/6009-0000001a", "RECMODE=dontcare") in new stack
    -- Executing [out@sub-record-check:3] ExecIf("SIP/6009-0000001a", "1?Goto(routewins)") in new stack
    -- Goto (sub-record-check,out,7)
    -- Executing [out@sub-record-check:7] Gosub("SIP/6009-0000001a", "recordcheck,1(dontcare,out,0531zzzzzz)") in new stack
    -- Executing [recordcheck@sub-record-check:1] NoOp("SIP/6009-0000001a", "Starting recording check against dontcare") in new stack
    -- Executing [recordcheck@sub-record-check:2] Goto("SIP/6009-0000001a", "dontcare") in new stack
    -- Goto (sub-record-check,recordcheck,3)
    -- Executing [recordcheck@sub-record-check:3] Return("SIP/6009-0000001a", "") in new stack
    -- Executing [out@sub-record-check:8] Return("SIP/6009-0000001a", "") in new stack
    -- Executing [0531zzzzzz@from-internal:3] Set("SIP/6009-0000001a", "MOHCLASS=atlantica-oldies") in new stack
    -- Executing [0531zzzzzz@from-internal:4] ExecIf("SIP/6009-0000001a", "0?Set(TRUNKCIDOVERRIDE=<0517392xxxxx>)") in new stack
    -- Executing [0531zzzzzz@from-internal:5] Set("SIP/6009-0000001a", "_NODEST=") in new stack
    -- Executing [0531zzzzzz@from-internal:6] Macro("SIP/6009-0000001a", "dialout-trunk,31,0531zzzzzz,,on") in new stack
    -- Executing [s@macro-dialout-trunk:1] Set("SIP/6009-0000001a", "DIAL_TRUNK=31") in new stack
    -- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/6009-0000001a", "0?sub-pincheck,s,1()") in new stack
    -- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/6009-0000001a", "0?disabletrunk,1") in new stack
    -- Executing [s@macro-dialout-trunk:4] Set("SIP/6009-0000001a", "DIAL_NUMBER=0531zzzzz") in new stack
    -- Executing [s@macro-dialout-trunk:5] Set("SIP/6009-0000001a", "DIAL_TRUNK_OPTIONS=Ttr") in new stack
    -- Executing [s@macro-dialout-trunk:6] Set("SIP/6009-0000001a", "OUTBOUND_GROUP=OUT_31") in new stack
    -- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/6009-0000001a", "0?nomax") in new stack
    -- Executing [s@macro-dialout-trunk:8] GotoIf("SIP/6009-0000001a", "0?chanfull") in new stack
    -- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/6009-0000001a", "0?skipoutcid") in new stack
    -- Executing [s@macro-dialout-trunk:10] Set("SIP/6009-0000001a", "DIAL_TRUNK_OPTIONS=T") in new stack
    -- Executing [s@macro-dialout-trunk:11] Macro("SIP/6009-0000001a", "outbound-callerid,31") in new stack
    -- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/6009-0000001a", "0?Set(CALLERPRES(name-pres)=)") in new stack
    -- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/6009-0000001a", "0?Set(CALLERPRES(num-pres)=)") in new stack
    -- Executing [s@macro-outbound-callerid:3] ExecIf("SIP/6009-0000001a", "0?Set(REALCALLERIDNUM=6009)") in new stack
    -- Executing [s@macro-outbound-callerid:4] GotoIf("SIP/6009-0000001a", "1?normcid") in new stack
    -- Goto (macro-outbound-callerid,s,7)
    -- Executing [s@macro-outbound-callerid:7] Set("SIP/6009-0000001a", "USEROUTCID="Name" <0517392xxxx>") in new stack
    -- Executing [s@macro-outbound-callerid:8] Set("SIP/6009-0000001a", "EMERGENCYCID=0517392xxxx") in new stack
    -- Executing [s@macro-outbound-callerid:9] Set("SIP/6009-0000001a", "TRUNKOUTCID=<0517392xxxx>") in new stack
    -- Executing [s@macro-outbound-callerid:10] GotoIf("SIP/6009-0000001a", "1?trunkcid") in new stack
    -- Goto (macro-outbound-callerid,s,15)
    -- Executing [s@macro-outbound-callerid:15] ExecIf("SIP/6009-0000001a", "1?Set(CALLERID(all)=<0517392xxxx>)") in new stack
    -- Executing [s@macro-outbound-callerid:16] ExecIf("SIP/6009-0000001a", "1?Set(CALLERID(all)="Name" <05173xxxxxx>)") in new stack
    -- Executing [s@macro-outbound-callerid:17] ExecIf("SIP/6009-0000001a", "1?Set(CALLERID(all)=<0517392xxxx>)") in new stack
    -- Executing [s@macro-outbound-callerid:18] ExecIf("SIP/6009-0000001a", "0?Set(CALLERPRES(name-pres)=prohib_passed_screen)") in new stack
    -- Executing [s@macro-outbound-callerid:19] ExecIf("SIP/6009-0000001a", "0?Set(CALLERPRES(num-pres)=prohib_passed_screen)") in new stack
    -- Executing [s@macro-outbound-callerid:20] Set("SIP/6009-0000001a", "CDR(outbound_cnum)=051739xxxxxx") in new stack
    -- Executing [s@macro-outbound-callerid:21] Set("SIP/6009-0000001a", "CDR(outbound_cnam)=") in new stack
    -- Executing [s@macro-dialout-trunk:12] GosubIf("SIP/6009-0000001a", "0?sub-flp-31,s,1()") in new stack
    -- Executing [s@macro-dialout-trunk:13] Set("SIP/6009-0000001a", "OUTNUM=0531zzzzz") in new stack
    -- Executing [s@macro-dialout-trunk:14] Set("SIP/6009-0000001a", "custom=SIP/telekom-92xxxx-chansip") in new stack
    -- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/6009-0000001a", "1?Set(DIAL_TRUNK_OPTIONS=M(setmusic^atlantica-oldies)T)") in new stack
    -- Executing [s@macro-dialout-trunk:16] ExecIf("SIP/6009-0000001a", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^atlantica-oldies)TM(confirm))") in new stack
    -- Executing [s@macro-dialout-trunk:17] Macro("SIP/6009-0000001a", "dialout-trunk-predial-hook,") in new stack
    -- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/6009-0000001a", "") in new stack
    -- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/6009-0000001a", "0?bypass,1") in new stack
    -- Executing [s@macro-dialout-trunk:19] ExecIf("SIP/6009-0000001a", "1?Set(CONNECTEDLINE(num,i)=0531zzzzzzzz)") in new stack
    -- Executing [s@macro-dialout-trunk:20] ExecIf("SIP/6009-0000001a", "1?Set(CONNECTEDLINE(name,i)=CID:05173xxxxxx)") in new stack
    -- Executing [s@macro-dialout-trunk:21] ExecIf("SIP/6009-0000001a", "0?Set(CONNECTEDLINE(name,i)=CID:(Hidden)05173xxxxxx)") in new stack
    -- Executing [s@macro-dialout-trunk:22] GotoIf("SIP/6009-0000001a", "0?customtrunk") in new stack
    -- Executing [s@macro-dialout-trunk:23] Dial("SIP/6009-0000001a", "SIP/telekom-925xxxxx-chansip/0531zzzzzz,300,M(setmusic^atlantica-oldies)T") in new stack
  == Using SIP VIDEO TOS bits 136
  == Using SIP VIDEO CoS mark 6
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
    -- Called SIP/telekom-92xxxxxx-chansip/0531zzzzzzz
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [s@macro-dialout-trunk:24] NoOp("SIP/6009-0000001a", "Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 58") in new stack
    -- Executing [s@macro-dialout-trunk:25] GotoIf("SIP/6009-0000001a", "1?continue,1:s-CHANUNAVAIL,1") in new stack
    -- Goto (macro-dialout-trunk,continue,1)
    -- Executing [continue@macro-dialout-trunk:1] NoOp("SIP/6009-0000001a", "TRUNK Dial failed due to CHANUNAVAIL HANGUPCAUSE: 58 - failing through to other trunks") in new stack


nur: raus wählen tut er auch nicht .....

(also, weder Telekom noch SIPGate noch Personal-VOiP - bluesip ganz am Ende der Kette tut es)

......
Code: 
    -- Executing [s@macro-dialout-trunk:22] GotoIf("SIP/6009-0000001a", "0?customtrunk") in new stack
    -- Executing [s@macro-dialout-trunk:23] Dial("SIP/6009-0000001a", "SIP/PersonalVoIP-2-chan_sip/0531zzzz,300,M(setmusic^atlantica-oldies)T") in new stack
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
    -- Called SIP/PersonalVoIP-2-chan_sip/0531zzzzzzzzz
[2019-01-08 15:46:29] WARNING[19331][C-0000002e]: chan_sip.c:24069 handle_response_invite: Received response: "Forbidden" from '"Name" <sip:[email protected]>;tag=as05e1aa83'
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [s@macro-dialout-trunk:24] NoOp("SIP/6009-0000001a", "Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 21") in new stack
    -- Executing [s@macro-dialout-trunk:25] GotoIf("SIP/6009-0000001a", "1?continue,1:s-CHANUNAVAIL,1") in new stack
    -- Goto (macro-dialout-trunk,continue,1)
    -- Executing [continue@macro-dialout-trunk:1] NoOp("SIP/6009-0000001a", "TRUNK Dial failed due to CHANUNAVAIL HANGUPCAUSE: 21 - failing through to other trunks") in new stack
    -- Executing [continue@macro-dialout-trunk:2] ExecIf("SIP/6009-0000001a", "1?Set(CALLERID(number)=6009)") in new stack
    -- Executing [0531zzzzzz@from-internal:12] Macro("SIP/6009-0000001a", "dialout-trunk,11,0531zzzzzzz,,off") in new stack
    -- Executing [s@macro-dialout-trunk:1] Set("SIP/6009-0000001a", "DIAL_TRUNK=11") in new stack
    -- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/6009-0000001a", "0?sub-pincheck,s,1()") in new stack
    -- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/6009-0000001a", "0?disabletrunk,1") in new stack
    -- Executing [s@macro-dialout-trunk:4] Set("SIP/6009-0000001a", "DIAL_NUMBER=0531zzzzzz") in new stack
    -- Executing [s@macro-dialout-trunk:5] Set("SIP/6009-0000001a", "DIAL_TRUNK_OPTIONS=Ttr") in new stack
    -- Executing [s@macro-dialout-trunk:6] Set("SIP/6009-0000001a", "OUTBOUND_GROUP=OUT_11") in new stack
    -- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/6009-0000001a", "0?nomax") in new stack
    -- Executing [s@macro-dialout-trunk:8] GotoIf("SIP/6009-0000001a", "0?chanfull") in new stack
    -- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/6009-0000001a", "0?skipoutcid") in new stack
    -- Executing [s@macro-dialout-trunk:10] Set("SIP/6009-0000001a", "DIAL_TRUNK_OPTIONS=T") in new stack
    -- Executing [s@macro-dialout-trunk:11] Macro("SIP/6009-0000001a", "outbound-callerid,11") in new stack
    -- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/6009-0000001a", "0?Set(CALLERPRES(name-pres)=)") in new stack
    -- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/6009-0000001a", "0?Set(CALLERPRES(num-pres)=)") in new stack
    -- Executing [s@macro-outbound-callerid:3] ExecIf("SIP/6009-0000001a", "0?Set(REALCALLERIDNUM=6009)") in new stack
    -- Executing [s@macro-outbound-callerid:4] GotoIf("SIP/6009-0000001a", "1?normcid") in new stack
    -- Goto (macro-outbound-callerid,s,7)
    -- Executing [s@macro-outbound-callerid:7] Set("SIP/6009-0000001a", "USEROUTCID="Name" <05173925xxxx>") in new stack
    -- Executing [s@macro-outbound-callerid:8] Set("SIP/6009-0000001a", "EMERGENCYCID=05173925xxxx") in new stack
    -- Executing [s@macro-outbound-callerid:9] Set("SIP/6009-0000001a", "TRUNKOUTCID=<004951734049939>") in new stack
    -- Executing [s@macro-outbound-callerid:10] GotoIf("SIP/6009-0000001a", "1?trunkcid") in new stack
    -- Goto (macro-outbound-callerid,s,15)
    -- Executing [s@macro-outbound-callerid:15] ExecIf("SIP/6009-0000001a", "1?Set(CALLERID(all)=<004951734049939>)") in new stack
    -- Executing [s@macro-outbound-callerid:16] ExecIf("SIP/6009-0000001a", "1?Set(CALLERID(all)="Name" <051739xxxx>)") in new stack
    -- Executing [s@macro-outbound-callerid:17] ExecIf("SIP/6009-0000001a", "1?Set(CALLERID(all)=<004951734049939>)") in new stack
    -- Executing [s@macro-outbound-callerid:18] ExecIf("SIP/6009-0000001a", "0?Set(CALLERPRES(name-pres)=prohib_passed_screen)") in new stack
    -- Executing [s@macro-outbound-callerid:19] ExecIf("SIP/6009-0000001a", "0?Set(CALLERPRES(num-pres)=prohib_passed_screen)") in new stack
    -- Executing [s@macro-outbound-callerid:20] Set("SIP/6009-0000001a", "CDR(outbound_cnum)=004951734049939") in new stack
    -- Executing [s@macro-outbound-callerid:21] Set("SIP/6009-0000001a", "CDR(outbound_cnam)=") in new stack
    -- Executing [s@macro-dialout-trunk:12] GosubIf("SIP/6009-0000001a", "0?sub-flp-11,s,1()") in new stack
    -- Executing [s@macro-dialout-trunk:13] Set("SIP/6009-0000001a", "OUTNUM=0531zzzzz") in new stack
    -- Executing [s@macro-dialout-trunk:14] Set("SIP/6009-0000001a", "custom=SIP/BlueSIP") in new stack
    -- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/6009-0000001a", "1?Set(DIAL_TRUNK_OPTIONS=M(setmusic^atlantica-oldies)T)") in new stack
    -- Executing [s@macro-dialout-trunk:16] ExecIf("SIP/6009-0000001a", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^atlantica-oldies)TM(confirm))") in new stack
    -- Executing [s@macro-dialout-trunk:17] Macro("SIP/6009-0000001a", "dialout-trunk-predial-hook,") in new stack
    -- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/6009-0000001a", "") in new stack
    -- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/6009-0000001a", "0?bypass,1") in new stack
    -- Executing [s@macro-dialout-trunk:19] ExecIf("SIP/6009-0000001a", "1?Set(CONNECTEDLINE(num,i)=0531zzzzzz)") in new stack
    -- Executing [s@macro-dialout-trunk:20] ExecIf("SIP/6009-0000001a", "1?Set(CONNECTEDLINE(name,i)=CID:00495173yyyyyyy)") in new stack
    -- Executing [s@macro-dialout-trunk:21] ExecIf("SIP/6009-0000001a", "0?Set(CONNECTEDLINE(name,i)=CID:(Hidden)00495173yyyyyyy)") in new stack
    -- Executing [s@macro-dialout-trunk:22] GotoIf("SIP/6009-0000001a", "0?customtrunk") in new stack
    -- Executing [s@macro-dialout-trunk:23] Dial("SIP/6009-0000001a", "SIP/BlueSIP/0531zzzzz,300,M(setmusic^atlantica-oldies)T") in new stack
  == Using SIP VIDEO TOS bits 136
  == Using SIP VIDEO CoS mark 6
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
    -- Called SIP/BlueSIP/0531zzzzzz
       > 0x6cc8a920 -- Strict RTP learning after remote address set to: 217.74.179.43:16720
    -- SIP/BlueSIP-00000021 is making progress passing it to SIP/6009-0000001a
  == Spawn extension (macro-dialout-trunk, s, 23) exited non-zero on 'SIP/6009-0000001a' in macro 'dialout-trunk'
  == Spawn extension (from-internal, 0531zzzzzz, 12) exited non-zero on 'SIP/6009-0000001a'


sngrep sagt dazu:

Call setup vom Softphone:

Code: 
    Call flow for i5ZYA4B-qVbVgqc5hb7QzQ.. (Color by Request/Response)
                                                            │INVITE sip:[email protected]:5060;transport=UDP SIP/2.0
             10.80.0.6:51616              192.168.80.12:5060│Via: SIP/2.0/UDP 10.80.0.6:51616;branch=z9hG4bK-524287-1---ad1e76515c930b35;rport
          ──────────┬─────────          ──────────┬─────────│Max-Forwards: 70
  15:52:08.405260   │        INVITE (SDP)         │         │Contact: <sip:[email protected]:51616;transport=UDP>
        +0.001614   │ ──────────────────────────> │         │To: <sip:[email protected]:5060;transport=UDP>
  15:52:08.406874   │      401 Unauthorized       │         │From: "Name"<sip:[email protected]:5060;transport=UDP>;tag=4f57322d
        +0.029201   │ <────────────────────────── │         │Call-ID: i5ZYA4B-qVbVgqc5hb7QzQ..
  15:52:08.436075   │             ACK             │         │CSeq: 1 INVITE
        +0.002357   │ ──────────────────────────> │         │Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
  15:52:08.438432   │        INVITE (SDP)         │         │Content-Type: application/sdp
        +0.027372   │ ──────────────────────────> │         │User-Agent: Zoiper rv2.9.2
  15:52:08.465804   │         100 Trying          │         │Allow-Events: presence, kpml, talk
        +2.104422   │ <────────────────────────── │         │Content-Length: 527
  15:52:10.570226   │  183 Session Progress (SDP) │         │
       +16.230942   │ <────────────────────────── │         │v=0
  15:52:26.801168   │        200 OK (SDP)         │         │o=Zoiper 122760384 0 IN IP4 10.80.0.6
        +0.099496   │ <────────────────────────── │         │s=Z
  15:52:26.900664   │        200 OK (SDP)         │         │c=IN IP4 10.80.0.6
        +0.200630   │ <<<──────────────────────── │         │t=0 0
  15:52:27.101294   │        200 OK (SDP)         │         │m=audio 32674 RTP/AVP 3 0 8 9 102 111 112 101 100 99
        +0.400249   │ <<<──────────────────────── │         │a=rtpmap:3 GSM/8000
  15:52:27.501543   │        200 OK (SDP)         │         │a=rtpmap:0 PCMU/8000
        +0.799282   │ <<<──────────────────────── │         │a=rtpmap:8 PCMA/8000
  15:52:28.300825   │        200 OK (SDP)         │         │a=rtpmap:9 G722/8000
        +1.600045   │ <<<──────────────────────── │         │a=rtpmap:102 G726-32/8000
  15:52:29.900870   │        200 OK (SDP)         │         │a=rtpmap:111 speex/16000
        +3.199735   │ <<<──────────────────────── │         │a=rtpmap:112 speex/32000
  15:52:33.100605   │        200 OK (SDP)         │         │a=rtpmap:101 telephone-event/8000
                    │ <<<──────────────────────── │         │a=fmtp:101 0-16
                    │                             │         │a=rtpmap:100 telephone-event/16000
                    │                             │         │a=fmtp:100 0-16
                    │                             │         │a=rtpmap:99 telephone-event/32000
                    │                             │         │a=fmtp:99 0-16
                    │                             │         │a=sendrecv
                    │                             │         │m=video 32676 RTP/AVP 116
                    │                             │         │a=rtpmap:116 VP8/90000
                    │                             │         │a=sendrecv
                    │                             │         │
                    │                             │         │
                    │                             │         │


Call zur Telekom:

Code: 
                                                                        Call flow for [email protected] (Color by Request/Response)
                                                            │INVITE sip:[email protected] SIP/2.0
            192.168.80.12:5060            217.0.20.192:5060 │Via: SIP/2.0/UDP 84.178.89.107:5060;branch=z9hG4bK19d79e05;rport
          ──────────┬─────────          ──────────┬─────────│Max-Forwards: 70
  15:52:08.758458   │        INVITE (SDP)         │         │From: <sip:[email protected]>;tag=as4aa83dec
        +0.072527   │ ──────────────────────────> │         │To: <sip:[email protected]>
  15:52:08.830985   │  407 Proxy Authentication R │         │Contact: <sip:[email protected]:5060>
        +0.000766   │ <────────────────────────── │         │Call-ID: [email protected]
  15:52:08.831751   │             ACK             │         │CSeq: 102 INVITE
        +0.001874   │ ──────────────────────────> │         │User-Agent: FPBX-13.0.192.19(13.22.0)
  15:52:08.833625   │        INVITE (SDP)         │         │Date: Tue, 08 Jan 2019 14:52:08 GMT
        +0.093491   │ ──────────────────────────> │         │Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  15:52:08.927116   │     606 Not Acceptable      │         │Supported: replaces, timer
        +0.000767   │ <────────────────────────── │         │Content-Type: application/sdp
  15:52:08.927883   │             ACK             │         │Content-Length: 1765
                    │ ──────────────────────────> │         │
                    │                             │         │v=0
                    │                             │         │o=root 139861392 139861392 IN IP4 84.178.89.107
                    │                             │         │s=Asterisk PBX 13.22.0
                    │                             │         │c=IN IP4 84.178.89.107
                    │                             │         │b=CT:384
                    │                             │         │t=0 0
                    │                             │         │m=audio 17364 RTP/AVP 9 0 8 3 111 4 110 116 18 7 97 107 112 101
                    │                             │         │a=rtpmap:9 G722/8000
                    │                             │         │a=rtpmap:0 PCMU/8000
                    │                             │         │a=rtpmap:8 PCMA/8000
                    │                             │         │a=rtpmap:3 GSM/8000


wie bitte, proxy authentification? Nanu?

Code: 
Proxy-Authorization: Digest username="e-mail ohne [USER=141086]@...[/USER]", realm="tel.t-online.de", algorithm=MD5, uri="sip:[email protected]", nonce="FE3E1CC924B9345C00000000320
        +0.093491   │ ──────────────────────────> │         │215", response="6cb86cd6273910d7464d856f19060d33", qop=auth, cnonce="55b4ddd7", nc=00000001
  15:52:08.927116   │     606 Not Acceptable      │         │Date: Tue, 08 Jan 2019 14:52:08 GMT

das mag er also nicht ...
SIP/2.0 606 Not Acceptable

vielleicht anders?

also Telefonnummer statt E-Mail:

Code: 
Proxy-Authorization: Digest username="05173aaaaaaa", realm="tel.t-online.de", algorithm=MD5, uri="sip:[email protected]", nonce="65D1363E1FBA345C000000004FB9FB06"
        +0.099635   │ ──────────────────────────> │         │response="de7c403b42ffb4f82c71b7fc91b33a4e", qop=auth, cnonce="2578dbe0", nc=00000001

nö, das mag er auch nicht

Code: 
SIP/2.0 606 Not Acceptable

die SIP settings dazu sind:

type=friend
username=05173aaaaaaa
fromuser=05173aaaaaaa
secret=password
host=tel.t-online.de
nat=yes
dtmfmode=rfc2833
canreinvite=update
fromdomain=tel.t-online.de
insecure=very
qualify=yes

05173aaaaaaa:password:[email protected]/92aaaaa

Ja, und nun gehen mir die Ideen aus ....

//edit by stoney: Bilder geschrumpft + [CODE] TAGs [/CODE] gesetzt
 

Anhänge

  • 2019-01-08 15_26_40-christian@DESKTOP-UERO69L_ ~.png
    2019-01-08 15_26_40-christian@DESKTOP-UERO69L_ ~.png
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  • 2019-01-08 15_39_35-Auswählen christian@DESKTOP-UERO69L_ ~.png
    2019-01-08 15_39_35-Auswählen christian@DESKTOP-UERO69L_ ~.png
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Zuletzt bearbeitet:
die Anschlußkennung dürfte da nirgends drin stehen ... die habe ich nirgends in der config vergraben?
 
Ich würde beim VoIP-Client erst einmal die Videoübertragung (zweite m-line) deaktivieren und die Codec-Liste für VoIP auf ein vernünftiges Maß (bspw. G722+G711a+u) reduzieren. Im Call-Flow des Softphones wird das 200OK nicht mit ACK bestätigt.
 
Von Softphone zu Softphone geht aber alles. Ebenso von Softphone zu MeetMe oder Softphone zu IVR.
Ich wüsste nicht, wie eine Einstellung im Softphone zB verhindert, dass ein eingehender Anruf von der Telekom überhaupt keine Spuren im Log hinterläßt ...

das scheint mir nicht die richtige Baustelle zu sein
 
Habe ich auch nicht behauptet, dass das die Probleme behebt. Trotzdem sollten hier die Hausaufgaben gemacht werden, um noch ganz andere Effekte vermeiden zu können.

Wie sieht denn das INVITE aus, auf das du das SIP606 oben bekommst? Ansonsten wären PCAPs von der Registrierung sowie ein- und ausgehenden Anrufen hilfreich, alternativ vollständige Offenlegung der SIP-Signalisierung (mit Anonymisierung). Wenn öffentliche PCAPs nicht möglich sind (bspw. nicht anonymisierbar), mir per PN schicken, dann würde ich mir das mal anschauen, ob etwas auffällig ist.
 
Für alle, die vielleicht ähnliche Probleme haben!

Outbound-Calls zur Telekom / All-IP:
  1. Im SDP darf kein Video-Codec enthalten sein.
  2. Alle Rufnummern im To / From / Request / Contact müssen im Format +E.164-Nummer sein. Also: +Länderkennung | Vorwahl ohne Null | Rufnummer. Bsp.: +4989234567 für eine Münchner Nummer z.B.
Dann wird das auch was mit dem Outbound call - wenn User und password korrekt sind, so dass die Authentifizierung erfolgreich ist. Aber wenn die nicht passt, gibt es auch keinen not acceptable sondern access denied oder ähnlich.

Wer auf Abbrüche von Calls nach z.B. 15 Minuten stößt, muss den pjsip-Treiber nutzen (würde ich sowieso grundsätzlich immer nehmen - chan_sip ist legacy).
 
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