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
bzw als zweites
und ganz zum Schluß:
In der Asterisk-Console (und auch im log-File) kommt davon NICHTS, kein einziges Zeichen an ...
sip show peers sagt aber:
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:
OK, das war nix.
Nächster Versuch:
diesmal gibt es was in der console:
nur: raus wählen tut er auch nicht .....
(also, weder Telekom noch SIPGate noch Personal-VOiP - bluesip ganz am Ende der Kette tut es)
......
sngrep sagt dazu:
Call setup vom Softphone:
Call zur Telekom:
wie bitte, proxy authentification? Nanu?
das mag er also nicht ...
SIP/2.0 606 Not Acceptable
vielleicht anders?
also Telefonnummer statt E-Mail:
nö, das mag er auch nicht
Ja, und nun gehen mir die Ideen aus ....
//edit by stoney: Bilder geschrumpft + [CODE] TAGs [/CODE] gesetzt
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
Zuletzt bearbeitet: