Probleme mit 1und1

Support
1

Hallöchen, wir sind von einem ISDN Anlagenanschluss auf einen 1und1 VOIP Account gewechselt.
Nun haben wir das Provlem, dass einegehende Anrufe nur sehr selten durchgestellt werden.
Raustelefonieren geht problemlos.

MobyDick sitzt hinter einer PFSense Firewall, bei der alle nötigen Ports per NAT an MD weitergeleitet werden.

Konfiguration:
Ämter: 1und1 mit Multi
Accounts (4 nur ein Beispiel):

Benutzername: 4991****
Passwort*****
Optionen:
fromuser=4991****
fromdomain=1und1.de
insecure=invite
nat=force_rport
qualify=yes
externhost=*****
allow=alaw

Typ: peer
Registrierung: Ja
Port: 5060
Durchwahl reg. 4991****
Ext. aus Header: Nein
Clip Modus: SIP Header

Peers sind alle online und die Registrierung klappt auch.

Eingehende Regel ist vorhanden und zum Testen auf “Catch All” gesetzt:
Bezeichnung: Standard
Quelle: *
Ziel: *
Durchwahl: 99
CID Name:
CID Nummer:
Sprache: deutsch

Sehr selten kommt mal ein Anruf durch. Meistens sieht die Fehlermeldung so aus:


-- Executing [4991****@no-auth-in:1] Gosub("SIP/1und1-3.sip.mgc.voip.telefonica.de-0000000f", "sub_emergency-check,s,1(4991****)") in new stack

-- Executing [s@sub_emergency-check:1] Verbose("SIP/1und1-3.sip.mgc.voip.telefonica.de-0000000f", "1,sub_emergency-check:: exten: 4991**** - descent: ") in new stack
 sub_emergency-check:: exten: 4991338258909 - descent:

-- Executing [s@sub_emergency-check:2] GotoIf("SIP/1und1-3.sip.mgc.voip.telefonica.de-0000000f", "1?4991****,1") in new stack
-- Goto (sub_emergency-check,4991****,1)
-- Channel 'SIP/1und1-3.sip.mgc.voip.telefonica.de-0000000f' sent to invalid extension: context,exten,priority=sub_emergency-check,4991****,1
-- Executing * Return("SIP/1und1-3.sip.mgc.voip.telefonica.de-0000000f", "") in new stack
-- Executing [4991****@no-auth-in:2] GotoIf("SIP/1und1-3.sip.mgc.voip.telefonica.de-0000000f", "0?mdc_emergency,dial,1:mdc_emergency,invalid,1") in new stack
-- Goto (mdc_emergency,invalid,1)
-- Executing [invalid@mdc_emergency:1] NoOp("SIP/1und1-3.sip.mgc.voip.telefonica.de-0000000f", "mdc_emergency::  is no emergency call") in new stack
-- Executing [invalid@mdc_emergency:2] Answer("SIP/1und1-3.sip.mgc.voip.telefonica.de-0000000f", "") in new stack
-- Executing [invalid@mdc_emergency:3] Playback("SIP/1und1-3.sip.mgc.voip.telefonica.de-0000000f", "beeperr") in new stack
-- <SIP/1und1-3.sip.mgc.voip.telefonica.de-0000000f> Playing 'beeperr.alaw' (language 'en')

[Nov  8 13:21:30] NOTICE[6986][C-0000000f]: channel.c:4257 __ast_read: Dropping incompatible voice frame on SIP/1und1-3.sip.mgc.voip.telefonica.de-0000000f of format ulaw since our native format has changed to (alaw)

-- Executing [invalid@mdc_emergency:4] Hangup("SIP/1und1-3.sip.mgc.voip.telefonica.de-0000000f", "19") in new stack
  == Spawn extension (mdc_emergency, invalid, 4) exited non-zero on 'SIP/1und1-3.sip.mgc.voip.telefonica.de-0000000f'

Jemand eine Idee?
Liebe Grüße und vielen Dank

Sven*

2

Hallo Sven

in deinem Log lese ich:
– Channel ‘SIP/1und1-3.sip.mgc.voip.telefonica.de-0000000f’ sent to invalid extension:

Ich denke Du hast (noch) ein Problem bei den eingehenden Regeln.

Grüße

Maik

3

Hallo Maik,

vielen Dank für Deine Antwort.

Eingehende Regel ist vorhanden und zum Testen auf “Catch All” gesetzt:
Bezeichnung: Standard
Quelle: *
Ziel: *
Durchwahl: 99
CID Name:
CID Nummer:
Sprache: deutsch

Auch wenn ich an diesen Einstellungen was verändere funktioniert es nicht.

Ich habe zum Test mal in den Systemeinstellungen bei sip.conf managed=0 eingestellt und manuell in der sip.conf bei General den Context auf mdc-incoming-23 geändert.
Zusätzlich noch no-auth deaktiviert.
Dann klappt es zumindest mit eingehenden Anrufen. Allerdings finde ich diese Lösung aus Sicherheitsgründen nicht so toll, weil dann auch ein paar Minuten später fraud/Spam Anrufe eingehen. :frowning:

Ich habe mir da schon ewig lange die Zähne ausgebissen.
Um ganz sicher zu gehen, habe ich auch mal die neue MobyDick Schule absolviert :slight_smile:

Was aber sehr merkwürdig ist:
In der Normaleinstellung kommen manchmal Anrufe durch.

Ich könnte mir vorstellen, dass es evtl. am Sipcluster von 1und1 liegt?
Im Internet liest man von zusätzlichen Einträgen in der sip.conf:

[1und1-1-2](1und1)
host=sipbalance1-2.1und1.de 

[1und1-2-1](1und1)
host=sipbalance2-1.1und1.de 

[1und1-2-2](1und1)
host=sipbalance2-2.1und1.de 

[1und1-3-1](1und1)
host=sipbalance3-1.1und1.de 

[1und1-3-2](1und1)
host=sipbalance3-2.1und1.de 

[1und1-4-1](1und1)
host=sipbalance4-1.1und1.de 

[1und1-4-2](1und1)
host=sipbalance4-2.1und1.de 

[1und1-5-1](1und1)
host=sipbalance5-1.1und1.de 

[1und1-5-2](1und1)
host=sipbalance5-2.1und1.de 

[1und1-6-1](1und1)
host=sipbalance6-1.1und1.de 

[1und1-6-2](1und1)
host=sipbalance6-2.1und1.de 

[1und1-7-1](1und1)
host=sipbalance7-1.1und1.de 

[1und1-7-2](1und1)
host=sipbalance7-2.1und1.de 

[1und1-8-1](1und1)
host=sipbalance8-1.1und1.de 

[1und1-8-2](1und1)
host=sipbalance8-2.1und1.de 

[telefonica-1](1und1)
host=1und1-1.sip.mgc.voip.telefonica.de

[telefonica-2](1und1)
host=1und1-2.sip.mgc.voip.telefonica.de

[telefonica-3](1und1)
host=1und1-3.sip.mgc.voip.telefonica.de

; [telefonica-4](1und1)
; host=1und1-4.sip.mgc.voip.telefonica.de    ; Habe ich noch nie gesehen

[telefonica-5](1und1)
host=1und1-5.sip.mgc.voip.telefonica.de

[telefonica-6](1und1)
host=1und1-6.sip.mgc.voip.telefonica.de

[telefonica-7](1und1)
host=1und1-7.sip.mgc.voip.telefonica.de

[telefonica-8](1und1)
host=1und1-8.sip.mgc.voip.telefonica.de

Ich denke nicht, dass man diese 1:1 so übernehmen kann, oder?
Ich bin leider mit meinem Latein am Ende. Ich vermisse den guten alten ISDN Anschluss…

Liebe Grüße

Sven

4

Hallo Sven,

in dem Thread hat schon jemand seine 1und1 Konfig gepostet, evtl. hilfts http://community.pascom.net/showthread.php?626-Probleme-mit-1und1&highlight=1und1.
Zu 1und1 gibt es noch mehr Einträge hier im Forum!

Grüße
Markus

5

Hallo Markus,

diese Beiträge habe ich alle schon durch.
Wie bereits geschrieben, habe ich keinerlei Probleme mit der Registrierung, etc.
Das Problem besteht darin, dass nur manchmal Anrufe durchkommen. Alle Fehlanrufe werden mit der o.g. Fehlermeldung quittiert.
Nach intensiver Beobachtung ist es so, dass alle Anrufe über Gateways von Telefonica zurückgewiesen werden.
Anrufe über die 1und1 Cluster kommen durch.

6

Ich habe mal einen Trace laufen lassen, während ein Anruf abgewiesen wurde.
Ich kann leider nichts entdecken, was auffällig wäre.
Jemand eine Idee?

  == Using SIP VIDEO TOS bits 136
  == Using SIP VIDEO CoS mark 6
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
    -- Executing [499133XXXXXXX@no-auth-in:1] Gosub("SIP/1und1-3.sip.mgc.voip.telefonica.de-00000015", "sub_emergency-check,s,1(499133XXXXXXX)") in new stack
    -- Executing [s@sub_emergency-check:1] Verbose("SIP/1und1-3.sip.mgc.voip.telefonica.de-00000015", "1,sub_emergency-check:: exten: 499133XXXXXXX - descent: ") in new stack
 sub_emergency-check:: exten: 499133XXXXXXX - descent:
    -- Executing [s@sub_emergency-check:2] GotoIf("SIP/1und1-3.sip.mgc.voip.telefonica.de-00000015", "1?499133XXXXXXX,1") in new stack
    -- Goto (sub_emergency-check,499133XXXXXXX,1)
    -- Channel 'SIP/1und1-3.sip.mgc.voip.telefonica.de-00000015' sent to invalid extension: context,exten,priority=sub_emergency-check,499133XXXXXXX,1
    -- Executing * Return("SIP/1und1-3.sip.mgc.voip.telefonica.de-00000015", "") in new stack
    -- Executing [499133XXXXXXX@no-auth-in:2] GotoIf("SIP/1und1-3.sip.mgc.voip.telefonica.de-00000015", "0?mdc_emergency,dial,1:mdc_emergency,invalid,1") in new stack
    -- Goto (mdc_emergency,invalid,1)
    -- Executing [invalid@mdc_emergency:1] NoOp("SIP/1und1-3.sip.mgc.voip.telefonica.de-00000015", "mdc_emergency::  is no emergency call") in new stack
    -- Executing [invalid@mdc_emergency:2] Answer("SIP/1und1-3.sip.mgc.voip.telefonica.de-00000015", "") in new stack
    -- Executing [invalid@mdc_emergency:3] Playback("SIP/1und1-3.sip.mgc.voip.telefonica.de-00000015", "beeperr") in new stack
    -- <SIP/1und1-3.sip.mgc.voip.telefonica.de-00000015> Playing 'beeperr.alaw' (language 'en')
[Nov 12 18:13:26] NOTICE[16052][C-00000014]: channel.c:4257 __ast_read: Dropping incompatible voice frame on SIP/1und1-3.sip.mgc.voip.telefonica.de-00000015 of format ulaw since our native format has changed to (alaw)
    -- Executing [invalid@mdc_emergency:4] Hangup("SIP/1und1-3.sip.mgc.voip.telefonica.de-00000015", "19") in new stack
  == Spawn extension (mdc_emergency, invalid, 4) exited non-zero on 'SIP/1und1-3.sip.mgc.voip.telefonica.de-00000015'
Reliably Transmitting (NAT) to 212.227.18.202:5060:
OPTIONS sip:sip.1und1.de SIP/2.0
Via: SIP/2.0/UDP 93.XXX.XXX.XXX:5060;branch=z9hG4bK119d7546;rport
Max-Forwards: 70
From: "asterisk" <sip:499133XXXXXXX@93.XXX.XXX.XXX>;tag=as7d3dc0b1
To: <sip:sip.1und1.de>
Contact: <sip:499133XXXXXXX@93.XXX.XXX.XXX:5060>
Call-ID: 1e18419c77daedf71961594e6af31eec@93.XXX.XXX.XXX:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.6-cert5
Date: Wed, 12 Nov 2014 17:13:32 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
Retransmitting #1 (NAT) to 212.227.18.202:5060:
OPTIONS sip:sip.1und1.de SIP/2.0
Via: SIP/2.0/UDP 93.XXX.XXX.XXX:5060;branch=z9hG4bK119d7546;rport
Max-Forwards: 70
From: "asterisk" <sip:499133XXXXXXX@93.XXX.XXX.XXX>;tag=as7d3dc0b1
To: <sip:sip.1und1.de>
Contact: <sip:499133XXXXXXX@93.XXX.XXX.XXX:5060>
Call-ID: 1e18419c77daedf71961594e6af31eec@93.206.202.161:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.6-cert5
Date: Wed, 12 Nov 2014 17:13:32 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
Retransmitting #2 (NAT) to 212.227.18.202:5060:
OPTIONS sip:sip.1und1.de SIP/2.0
Via: SIP/2.0/UDP 93.XXX.XXX.XXX:5060;branch=z9hG4bK119d7546;rport
Max-Forwards: 70
From: "asterisk" <sip:499133XXXXXXX@93.XXX.XXX.XXX>;tag=as7d3dc0b1
To: <sip:sip.1und1.de>
Contact: <sip:499133XXXXXXX@93.XXX.XXX.XXX:5060>
Call-ID: 1e18419c77daedf71961594e6af31eec@93.XXX.XXX.XXX:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.6-cert5
Date: Wed, 12 Nov 2014 17:13:32 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
Retransmitting #3 (NAT) to 212.227.18.202:5060:
OPTIONS sip:sip.1und1.de SIP/2.0
Via: SIP/2.0/UDP 93.XXX.XXX.XXX:5060;branch=z9hG4bK119d7546;rport
Max-Forwards: 70
From: "asterisk" <sip:499133XXXXXXX@93.XXX.XXX.XXX>;tag=as7d3dc0b1
To: <sip:sip.1und1.de>
Contact: <sip:499133XXXXXXX@93.XXX.XXX.XXX:5060>
Call-ID: 1e18419c77daedf71961594e6af31eec@93.206.202.161:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.6-cert5
Date: Wed, 12 Nov 2014 17:13:32 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
Retransmitting #4 (NAT) to 212.227.18.202:5060:
OPTIONS sip:sip.1und1.de SIP/2.0
Via: SIP/2.0/UDP 93.XXX.XXX.XXX:5060;branch=z9hG4bK119d7546;rport
Max-Forwards: 70
From: "asterisk" <sip:499133XXXXXXX@93.XXX.XXX.XXX>;tag=as7d3dc0b1
To: <sip:sip.1und1.de>
Contact: <sip:499133XXXXXXX@93.XXX.XXX.XXX:5060>
Call-ID: 1e18419c77daedf71961594e6af31eec@93.XXX.XXX.XXX:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.6-cert5
Date: Wed, 12 Nov 2014 17:13:32 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
Really destroying SIP dialog '1e18419c77daedf71961594e6af31eec@93.XXX.XXX.XXX:5060' Method: OPTIONS
mobydick*CLI>
Disconnected from Asterisk server
Asterisk cleanly ending (0).
Executing last minute cleanups
```*
7

Guten Abend,

schau dir mal den Thread hierzu an http://community.pascom.net/showthread.php?948-SID-Peering-Deutsche-Telefon&highlight=auth.

Grüße
Markus

8

Hallo Markus,

wenn ich den von Dir erwähnten Beitrag richtig interpretiere, ist er nicht wirklich eine Lösung für mein Problem.

Ich kann ja raustelefonieren. Eingehende Anrufe klappen. Aber nur sehr selten.
Das denke ich liegt daran, dass sip.1und1.de einen Cluster verwendet und die Anrufe dann nicht von der IP kommen, mit der sich der entsprechende Trunk registriert hat.
Sollte der Anruf durch Zufall von der im Trunk registrierten IP kommen geht es:

  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
    -- Executing [499133XXXXXXX@mdc_incoming-23:1] Set("SIP/mdc_trunk_conf-23-0000001a", "MDC_CALLER_NUM_TRUNK=+49151XXXXXXXX") in new stack
    -- Executing [499133XXXXXXX@mdc_incoming-23:2] Set("SIP/mdc_trunk_conf-23-0000001a", "MDC_CALLEE_NUM_TRUNK=499133XXXXXXX") in new stack
    -- Executing [499133XXXXXXX@mdc_incoming-23:3] Goto("SIP/mdc_trunk_conf-23-0000001a", "mdc_trunk-22,s,1") in new stack
    -- Goto (mdc_trunk-22,s,1)
    -- Executing [s@mdc_trunk-22:1] Verbose("SIP/mdc_trunk_conf-23-0000001a", "1,callee number: 499133XXXXXXX caller number: +49151XXXXXXXX") in new stack
 callee number: 499133XXXXXXX caller number: +49151XXXXXXXX
    -- Executing [s@mdc_trunk-22:2] Set("SIP/mdc_trunk_conf-23-0000001a", "MDC_FALLBACK_TRUNK=99") in new stack
    -- Executing [s@mdc_trunk-22:3] Gosub("SIP/mdc_trunk_conf-23-0000001a", "sub_nat2int,s,1(MDC_CALLER_NUM_INTERNAT,+49151XXXXXXXX,00,49,0,9133)") in new stack
    -- Executing [s@sub_nat2int:1] Verbose("SIP/mdc_trunk_conf-23-0000001a", "1,sub_nat2int:: variable: MDC_CALLER_NUM_INTERNAT - number: +49151XXXXXXXX - intVAZ: 00 - lkz: 49 - natVAZ: 0 - onKz: 9133") in new stack
 sub_nat2int:: variable: MDC_CALLER_NUM_INTERNAT - number: +49151XXXXXXXX - intVAZ: 00 - lkz: 49 - natVAZ: 0 - onKz: 9133
    -- Executing [s@sub_nat2int:2] GotoIf("SIP/mdc_trunk_conf-23-0000001a", "0?s-emergency,1") in new stack
    -- Executing [s@sub_nat2int:3] GotoIf("SIP/mdc_trunk_conf-23-0000001a", "0?s-int,1") in new stack
    -- Executing [s@sub_nat2int:4] GotoIf("SIP/mdc_trunk_conf-23-0000001a", "0?s-int,1") in new stack
    -- Executing [s@sub_nat2int:5] GotoIf("SIP/mdc_trunk_conf-23-0000001a", " 1?s-convert,1") in new stack
    -- Goto (sub_nat2int,s-convert,1)
    -- Executing [s-convert@sub_nat2int:1] Set("SIP/mdc_trunk_conf-23-0000001a", "ARG2=0049151XXXXXXXX") in new stack
    -- Executing [s-convert@sub_nat2int:2] Goto("SIP/mdc_trunk_conf-23-0000001a", "s,check") in new stack
    -- Goto (sub_nat2int,s,6)
    -- Executing [s@sub_nat2int:6] GotoIf("SIP/mdc_trunk_conf-23-0000001a", "1?s-int,1") in new stack
    -- Goto (sub_nat2int,s-int,1)
    -- Executing [s-int@sub_nat2int:1] Verbose("SIP/mdc_trunk_conf-23-0000001a", "1,international") in new stack
 international
    -- Executing [s-int@sub_nat2int:2] Set("SIP/mdc_trunk_conf-23-0000001a", "MDC_CALLER_NUM_INTERNAT=0049151XXXXXXXX") in new stack
    -- Executing [s-int@sub_nat2int:3] Return("SIP/mdc_trunk_conf-23-0000001a", "") in new stack
    -- Executing [s@mdc_trunk-22:4] Set("SIP/mdc_trunk_conf-23-0000001a", "CALLERID(num)=0049151XXXXXXXX") in new stack
    -- Executing [s@mdc_trunk-22:5] Gosub("SIP/mdc_trunk_conf-23-0000001a", "sub_int2nat,s,1(MDC_CALLER_NUM_NAT,0049151XXXXXXXX,00,49,0,9133)") in new stack
    -- Executing [s@sub_int2nat:1] Verbose("SIP/mdc_trunk_conf-23-0000001a", "1,sub_int2nat:: variable: MDC_CALLER_NUM_NAT - exten: 0049151XXXXXXXX - intVAZ: 00 - lkz: 49 - natVAZ: 0 - onKz: 9133") in new stack
 sub_int2nat:: variable: MDC_CALLER_NUM_NAT - exten: 0049151XXXXXXXX - intVAZ: 00 - lkz: 49 - natVAZ: 0 - onKz: 9133
    -- Executing [s@sub_int2nat:2] GotoIf("SIP/mdc_trunk_conf-23-0000001a", "0?s-emergency,1") in new stack
    -- Executing [s@sub_int2nat:3] GotoIf("SIP/mdc_trunk_conf-23-0000001a", "0?nat") in new stack
    -- Executing [s@sub_int2nat:4] GotoIf("SIP/mdc_trunk_conf-23-0000001a", "1?s-internat,1") in new stack
    -- Goto (sub_int2nat,s-internat,1)
    -- Executing [s-internat@sub_int2nat:1] Set("SIP/mdc_trunk_conf-23-0000001a", "MDC_CALLER_NUM_NAT=0151XXXXXXXX") in new stack
    -- Executing [s-internat@sub_int2nat:2] Return("SIP/mdc_trunk_conf-23-0000001a", "") in new stack
    -- Executing [s@mdc_trunk-22:6] Set("SIP/mdc_trunk_conf-23-0000001a", "CALLERID(num)=0151XXXXXXXX") in new stack
    -- Executing [s@mdc_trunk-22:7] UserEvent("SIP/mdc_trunk_conf-23-0000001a", "ResolveCallerName,Strategy: default,Outbound: 0,Channel: SIP/mdc_trunk_conf-23-0000001a") in new stack
    -- Executing [s@mdc_trunk-22:8] Wait("SIP/mdc_trunk_conf-23-0000001a", "0.25") in new stack
    -- Executing [s@mdc_trunk-22:9] Verbose("SIP/mdc_trunk_conf-23-0000001a", "1,MDC_RESOLVENAME_HITS = 1") in new stack
 MDC_RESOLVENAME_HITS = 1
    -- Executing [s@mdc_trunk-22:10] Verbose("SIP/mdc_trunk_conf-23-0000001a", "1,CALLERID(name) = Sven") in new stack
 CALLERID(name) = Sven
    -- Executing [s@mdc_trunk-22:11] Set("SIP/mdc_trunk_conf-23-0000001a", "MDC_NUMPREFIX_TRUNK=0") in new stack
    -- Executing [s@mdc_trunk-22:12] ExecIf("SIP/mdc_trunk_conf-23-0000001a", "1?Set(CALLERID(num)=00151XXXXXXXX)") in new stack
    -- Executing [s@mdc_trunk-22:13] Goto("SIP/mdc_trunk_conf-23-0000001a", "mdc_mapping-22,499133XXXXXXX,1") in new stack
    -- Goto (mdc_mapping-22,499133XXXXXXX,1)
    -- Executing [499133XXXXXXX@mdc_mapping-22:1] Set("SIP/mdc_trunk_conf-23-0000001a", "CHANNEL(language)=de") in new stack
    -- Executing [499133XXXXXXX@mdc_mapping-22:2] Verbose("SIP/mdc_trunk_conf-23-0000001a", "1,mapping from _*%#a-zA-Z0-9]. to 99") in new stack
 mapping from _*%#a-zA-Z0-9]. to 99
    -- Executing [499133XXXXXXX@mdc_mapping-22:3] Goto("SIP/mdc_trunk_conf-23-0000001a", "mdc_external,99,1") in new stack
    -- Goto (mdc_external,99,1)
    -- Executing [99@mdc_external:1] SIPAddHeader("SIP/mdc_trunk_conf-23-0000001a", ""Alert-Info:<http://www.notused.de>;info=alert-external;x-line-id=0"") in new stack
    -- Executing [99@mdc_external:2] GosubIf("SIP/mdc_trunk_conf-23-0000001a", "1?sub_initcall,s,1(ext,99)") in new stack
    -- Executing [s@sub_initcall:1] Verbose("SIP/mdc_trunk_conf-23-0000001a", "1,sub_initcall descent: ext exten: 99") in new stack
 sub_initcall descent: ext exten: 99
    -- Executing [s@sub_initcall:2] GosubIf("SIP/mdc_trunk_conf-23-0000001a", "1?sub_initloop,s,1") in new stack
    -- Executing [s@sub_initloop:1] Verbose("SIP/mdc_trunk_conf-23-0000001a", "1,initial loop") in new stack
 initial loop
    -- Executing [s@sub_initloop:2] Set("SIP/mdc_trunk_conf-23-0000001a", "MDC_ALIAS_HOP=0") in new stack
    -- Executing [s@sub_initloop:3] Return("SIP/mdc_trunk_conf-23-0000001a", "") in new stack
    -- Executing [s@sub_initcall:3] Set("SIP/mdc_trunk_conf-23-0000001a", "__MDC_TRANSFERBACK_HOP=0") in new stack
    -- Executing [s@sub_initcall:4] Set("SIP/mdc_trunk_conf-23-0000001a", "__MDC_DIALDESCENT=ext") in new stack
    -- Executing [s@sub_initcall:5] Goto("SIP/mdc_trunk_conf-23-0000001a", "ext,1") in new stack
    -- Goto (sub_initcall,ext,1)
    -- Executing [ext@sub_initcall:1] Set("SIP/mdc_trunk_conf-23-0000001a", "__MDC_DIALCALLERNUMINIT=+49151XXXXXXXX") in new stack
    -- Executing [ext@sub_initcall:2] Set("SIP/mdc_trunk_conf-23-0000001a", "__MDC_DIALCALLEENUMINIT=499133XXXXXXX") in new stack
    -- Executing [ext@sub_initcall:3] Set("SIP/mdc_trunk_conf-23-0000001a", "__MDC_DIALCALLEENUMMAP=99") in new stack
    -- Executing [ext@sub_initcall:4] Return("SIP/mdc_trunk_conf-23-0000001a", "") in new stack
    -- Executing [99@mdc_external:3] Goto("SIP/mdc_trunk_conf-23-0000001a", "main,99,1") in new stack
    -- Goto (main,99,1)
    -- Executing [99@main:1] Gosub("SIP/mdc_trunk_conf-23-0000001a", "sub_defcall,s,1(99)") in new stack
    -- Executing [s@sub_defcall:1] Set("SIP/mdc_trunk_conf-23-0000001a", "__MDC_ALIAS_HOP=1") in new stack
    -- Executing [s@sub_defcall:2] Set("SIP/mdc_trunk_conf-23-0000001a", "__MDC_DIALCALLEENUM=99") in new stack
    -- Executing [s@sub_defcall:3] Set("SIP/mdc_trunk_conf-23-0000001a", "__MDC_DIALCALLERNUM=00151XXXXXXXX") in new stack
    -- Executing [s@sub_defcall:4] GotoIf("SIP/mdc_trunk_conf-23-0000001a", "1?nozap") in new stack
    -- Goto (sub_defcall,s,8)
    -- Executing [s@sub_defcall:8] Set("SIP/mdc_trunk_conf-23-0000001a", "__MDC_DIALCALLERNAME=Sven") in new stack
    -- Executing [s@sub_defcall:9] Set("SIP/mdc_trunk_conf-23-0000001a", "__MDC_DIALCHANNELNAME=mdc_trunk_conf-23") in new stack
    -- Executing [s@sub_defcall:10] Return("SIP/mdc_trunk_conf-23-0000001a", "") in new stack
    -- Executing [99@main:2] Goto("SIP/mdc_trunk_conf-23-0000001a", "mdc_distribute,99,1") in new stack
    -- Goto (mdc_distribute,99,1)
    -- Executing [99@mdc_distribute:1] Gosub("SIP/mdc_trunk_conf-23-0000001a", "sub_user,s,1(callee,7,sven,Sven,99,99)") in new stack
    -- Executing [s@sub_user:1] Verbose("SIP/mdc_trunk_conf-23-0000001a", "1,sub_user mode callee") in new stack
 sub_user mode callee
    -- Executing [s@sub_user:2] Goto("SIP/mdc_trunk_conf-23-0000001a", "s-callee,1") in new stack
    -- Goto (sub_user,s-callee,1)
    -- Executing [s-callee@sub_user:1] Verbose("SIP/mdc_trunk_conf-23-0000001a", "1,sub_user user id: 7 - user: sven - realname: Sven - own mailbox: 99 - watched mailbox: 99") in new stack
 sub_user user id: 7 - user: sven - realname: Sven - own mailbox: 99 - watched mailbox: 99
    -- Executing [s-callee@sub_user:2] Set("SIP/mdc_trunk_conf-23-0000001a", "_MDC_CALLEE_USER_ID=7") in new stack
    -- Executing [s-callee@sub_user:3] Set("SIP/mdc_trunk_conf-23-0000001a", "_MDC_CALLEE_ACC_NAME=sven") in new stack
    -- Executing [s-callee@sub_user:4] Set("SIP/mdc_trunk_conf-23-0000001a", "_MDC_CALLEE_ACC_REALNAME=Sven") in new stack
    -- Executing [s-callee@sub_user:5] Set("SIP/mdc_trunk_conf-23-0000001a", "_MDC_CALLEE_VM_OWN=99") in new stack
    -- Executing [s-callee@sub_user:6] Set("SIP/mdc_trunk_conf-23-0000001a", "_MDC_CALLEE_VM_WATCHED=99") in new stack
    -- Executing [s-callee@sub_user:7] Return("SIP/mdc_trunk_conf-23-0000001a", "") in new stack
    -- Executing [99@mdc_distribute:2] Goto("SIP/mdc_trunk_conf-23-0000001a", "mdc_ident-7,99,1") in new stack
    -- Goto (mdc_ident-7,99,1)
    -- Executing [99@mdc_ident-7:1] NoOp("SIP/mdc_trunk_conf-23-0000001a", "alias-check:: call forwarding from 99 - 0") in new stack
    -- Executing [99@mdc_ident-7:2] GotoIf("SIP/mdc_trunk_conf-23-0000001a", "0?99-uml,1:99-dial,1") in new stack
    -- Goto (mdc_ident-7,99-dial,1)
    -- Executing [99-dial@mdc_ident-7:1] Set("SIP/mdc_trunk_conf-23-0000001a", "__MDC_EXTEN=99") in new stack
    -- Executing [99-dial@mdc_ident-7:2] Gosub("SIP/mdc_trunk_conf-23-0000001a", "sub_prefix-99,ext,1") in new stack
    -- Executing [ext@sub_prefix-99:1] Verbose("SIP/mdc_trunk_conf-23-0000001a", "1,no action") in new stack
 no action
    -- Executing [ext@sub_prefix-99:2] Return("SIP/mdc_trunk_conf-23-0000001a", "") in new stack
    -- Executing [99-dial@mdc_ident-7:3] GotoIf("SIP/mdc_trunk_conf-23-0000001a", "0?99-unavail,1") in new stack
    -- Executing [99-dial@mdc_ident-7:4] GotoIf("SIP/mdc_trunk_conf-23-0000001a", "0?99-busy,1") in new stack
    -- Executing [99-dial@mdc_ident-7:5] Gosub("SIP/mdc_trunk_conf-23-0000001a", "sub_main-99,ext,1") in new stack
    -- Executing [ext@sub_main-99:1] Set("SIP/mdc_trunk_conf-23-0000001a", "__PICKUPMARK=99") in new stack
    -- Executing [ext@sub_main-99:2] Set("SIP/mdc_trunk_conf-23-0000001a", "__SPYGROUP=99") in new stack
    -- Executing [ext@sub_main-99:3] Dial("SIP/mdc_trunk_conf-23-0000001a", "Local/99@mdc_diallocation") in new stack
    -- Called Local/99@mdc_diallocation
    -- Executing [99@mdc_diallocation:1] Verbose("Local/99@mdc_diallocation-0000001c;2", "1,dial to 99 for user 7") in new stack
 dial to 99 for user 7
    -- Executing [99@mdc_diallocation:2] Set("Local/99@mdc_diallocation-0000001c;2", "__MDC_DIALLOCATION_CHANNEL=Local/99@mdc_diallocation-0000001c;2") in new stack
    -- Executing [99@mdc_diallocation:3] Set("Local/99@mdc_diallocation-0000001c;2", "SHARED(MDC_CALLEE_PEERS)=") in new stack
    -- Executing [99@mdc_diallocation:4] Set("Local/99@mdc_diallocation-0000001c;2", "MDC_EXTEN=99") in new stack
    -- Executing [99@mdc_diallocation:5] ExecIf("Local/99@mdc_diallocation-0000001c;2", "0?Set(TMP_TARGET=team):Set(TMP_TARGET=ext)") in new stack
    -- Executing [99@mdc_diallocation:6] Gosub("Local/99@mdc_diallocation-0000001c;2", "sub_split-user,s,1(7)") in new stack
    -- Executing [s@sub_split-user:1] Verbose("Local/99@mdc_diallocation-0000001c;2", "1,get locations for for: 7") in new stack
 get locations for for: 7
    -- Executing [s@sub_split-user:2] GotoIf("Local/99@mdc_diallocation-0000001c;2", "0?s-zero,1") in new stack
    -- Executing [s@sub_split-user:3] Set("Local/99@mdc_diallocation-0000001c;2", "TMP_LOCATIONS=3;0") in new stack
    -- Executing [s@sub_split-user:4] Set("Local/99@mdc_diallocation-0000001c;2", "TMP_STATIC_ID=3") in new stack
    -- Executing [s@sub_split-user:5] Verbose("Local/99@mdc_diallocation-0000001c;2", "1,static id: 3") in new stack
 static id: 3
    -- Executing [s@sub_split-user:6] Set("Local/99@mdc_diallocation-0000001c;2", "TMP_DYN_ID=0") in new stack
    -- Executing [s@sub_split-user:7] Verbose("Local/99@mdc_diallocation-0000001c;2", "1,dynamic id: 0") in new stack
 dynamic id: 0
    -- Executing [s@sub_split-user:8] Return("Local/99@mdc_diallocation-0000001c;2", "") in new stack
    -- Executing [99@mdc_diallocation:7] GotoIf("Local/99@mdc_diallocation-0000001c;2", "0?invalid,1") in new stack
    -- Executing [99@mdc_diallocation:8] GotoIf("Local/99@mdc_diallocation-0000001c;2", "0?dynamic,1") in new stack
    -- Executing [99@mdc_diallocation:9] GotoIf("Local/99@mdc_diallocation-0000001c;2", "1?static,1") in new stack
    -- Goto (mdc_diallocation,static,1)
    -- Executing [static@mdc_diallocation:1] Dial("Local/99@mdc_diallocation-0000001c;2", "Local/ext@mdc_locallocation-3") in new stack
    -- Called Local/ext@mdc_locallocation-3
    -- Executing [ext@mdc_locallocation-3:1] Set("Local/ext@mdc_locallocation-3-0000001d;2", "PUSH(SHARED(MDC_CALLEE_PEERS,Local/99@mdc_diallocation-0000001c;2))=Local/ext-1@mdc_localdevice-3") in new stack
    -- Executing [ext@mdc_locallocation-3:2] Set("Local/ext@mdc_locallocation-3-0000001d;2", "PUSH(SHARED(MDC_CALLEE_PEERS,Local/99@mdc_diallocation-0000001c;2))=Local/ext-7@mdc_localdevice-3") in new stack
    -- Executing [ext@mdc_locallocation-3:3] Set("Local/ext@mdc_locallocation-3-0000001d;2", "PUSH(SHARED(MDC_CALLEE_PEERS,Local/99@mdc_diallocation-0000001c;2))=Local/ext-3@mdc_localdevice-3") in new stack
    -- Executing [ext@mdc_locallocation-3:4] Set("Local/ext@mdc_locallocation-3-0000001d;2", "PUSH(SHARED(MDC_CALLEE_PEERS,Local/99@mdc_diallocation-0000001c;2))=Local/ext-4@mdc_localdevice-3") in new stack
    -- Executing [ext@mdc_locallocation-3:5] Dial("Local/ext@mdc_locallocation-3-0000001d;2", "Local/ext-1@mdc_localdevice-3&Local/ext-7@mdc_localdevice-3&Local/ext-3@mdc_localdevice-3&Local/ext-4@mdc_localdevice-3") in new stack
    -- Called Local/ext-1@mdc_localdevice-3
    -- Called Local/ext-7@mdc_localdevice-3
    -- Called Local/ext-3@mdc_localdevice-3
    -- Called Local/ext-4@mdc_localdevice-3
    -- Executing [ext-1@mdc_localdevice-3:1] Verbose("Local/ext-1@mdc_localdevice-3-0000001e;2", "1,Dialing with delay of 0 seconds for 99 seconds") in new stack
 Dialing with delay of 0 seconds for 99 seconds
    -- Executing [ext-1@mdc_localdevice-3:2] Dial("Local/ext-1@mdc_localdevice-3-0000001e;2", "SIP/NcEod2OoIdRmlQj,99") in new stack
[Nov 13 10:27:49] WARNING[8224][C-00000014]: app_dial.c:2437 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [ext-1@mdc_localdevice-3:3] Verbose("Local/ext-1@mdc_localdevice-3-0000001e;2", "1,mdc_localdevice-Dialstatus: CHANUNAVAIL - cause: 20") in new stack
 mdc_localdevice-Dialstatus: CHANUNAVAIL - cause: 20
    -- Executing [ext-1@mdc_localdevice-3:4] Gosub("Local/ext-1@mdc_localdevice-3-0000001e;2", "sub_hangup,s,1(CHANUNAVAIL)") in new stack
    -- Executing [s@sub_hangup:1] Verbose("Local/ext-1@mdc_localdevice-3-0000001e;2", "1,sub_hangup dialstatus: CHANUNAVAIL") in new stack
 sub_hangup dialstatus: CHANUNAVAIL
    -- Executing [s@sub_hangup:2] GotoIf("Local/ext-1@mdc_localdevice-3-0000001e;2", "0?noanswer,1") in new stack
    -- Executing [s@sub_hangup:3] GotoIf("Local/ext-1@mdc_localdevice-3-0000001e;2", "0?busy,1") in new stack
    -- Executing [s@sub_hangup:4] Goto("Local/ext-1@mdc_localdevice-3-0000001e;2", "unavailable,1") in new stack
    -- Goto (sub_hangup,unavailable,1)
    -- Executing [unavailable@sub_hangup:1] Hangup("Local/ext-1@mdc_localdevice-3-0000001e;2", "20") in new stack
  == Spawn extension (sub_hangup, unavailable, 1) exited non-zero on 'Local/ext-1@mdc_localdevice-3-0000001e;2'
    -- Executing [ext-7@mdc_localdevice-3:1] Verbose("Local/ext-7@mdc_localdevice-3-0000001f;2", "1,Dialing with delay of 0 seconds for 999 seconds") in new stack
 Dialing with delay of 0 seconds for 999 seconds
    -- Executing [ext-7@mdc_localdevice-3:2] Dial("Local/ext-7@mdc_localdevice-3-0000001f;2", "SIP/asus,999") in new stack
[Nov 13 10:27:49] WARNING[8225][C-00000014]: app_dial.c:2437 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [ext-7@mdc_localdevice-3:3] Verbose("Local/ext-7@mdc_localdevice-3-0000001f;2", "1,mdc_localdevice-Dialstatus: CHANUNAVAIL - cause: 20") in new stack
 mdc_localdevice-Dialstatus: CHANUNAVAIL - cause: 20
    -- Executing [ext-7@mdc_localdevice-3:4] Gosub("Local/ext-7@mdc_localdevice-3-0000001f;2", "sub_hangup,s,1(CHANUNAVAIL)") in new stack
    -- Executing [s@sub_hangup:1] Verbose("Local/ext-7@mdc_localdevice-3-0000001f;2", "1,sub_hangup dialstatus: CHANUNAVAIL") in new stack
 sub_hangup dialstatus: CHANUNAVAIL
    -- Executing [s@sub_hangup:2] GotoIf("Local/ext-7@mdc_localdevice-3-0000001f;2", "0?noanswer,1") in new stack
    -- Executing [s@sub_hangup:3] GotoIf("Local/ext-7@mdc_localdevice-3-0000001f;2", "0?busy,1") in new stack
    -- Executing [s@sub_hangup:4] Goto("Local/ext-7@mdc_localdevice-3-0000001f;2", "unavailable,1") in new stack
    -- Goto (sub_hangup,unavailable,1)
    -- Executing [unavailable@sub_hangup:1] Hangup("Local/ext-7@mdc_localdevice-3-0000001f;2", "20") in new stack
  == Spawn extension (sub_hangup, unavailable, 1) exited non-zero on 'Local/ext-7@mdc_localdevice-3-0000001f;2'
    -- Executing [ext-3@mdc_localdevice-3:1] Verbose("Local/ext-3@mdc_localdevice-3-00000020;2", "1,Dialing with delay of 0 seconds for 99 seconds") in new stack
 Dialing with delay of 0 seconds for 99 seconds
    -- Executing [ext-3@mdc_localdevice-3:2] Dial("Local/ext-3@mdc_localdevice-3-00000020;2", "SIP/creadoo1,99") 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/creadoo1
    -- Executing [ext-4@mdc_localdevice-3:1] Verbose("Local/ext-4@mdc_localdevice-3-00000021;2", "1,Dialing with delay of 0 seconds for 99 seconds") in new stack
 Dialing with delay of 0 seconds for 99 seconds


...usw.

Besteht die Möglichkeit der MD irgendwie die verschiedenen 1und1 Adressen mitzuteilen?

9

Re,

hast du schon mal versucht mit tcpdump bzw. sipgrep rauszufiltern, von welchen HOSTs die eingehenden Gespräche kommen?
Falls diese wie du sagst abweichen, könntest du mal versuchen für jeden unterschiedlichen HOST ein Account für das Amt zu erstellen, kontrolliere dann bitte, dass in der sip.conf der host=xxxx richtig drin steht, bzw. unter den Optionen mit anzugeben.

Grüße
Markus

10

Das werde ich mal versuchen.
Gibt es alternativ die Möglichkeit auch einen Netzwerkbereich anzugeben (z.B. 212.227.0.0/16) ?

11

Hier mal der sipgrep Dump:

root@mobydick:/etc/admin# sipgrep
interface: eth0 (192.168.10.0/255.255.255.0)
filter: (ip) and ( portrange 5060-5061)

U 2014/11/13 13:30:56.358113 212.227.18.134:5060 -> 192.168.10.207:5060
INVITE sip:499133XXXXXXX@93.206.XXX.XXX:5060 SIP/2.0.
Record-Route: <sip:212.227.18.134;lr=on>.
Record-Route: <sip:212.227.18.164;lr=on;did=89a.418e8ff>.
Record-Route: <sip:212.227.18.199;lr=on>.
Via: SIP/2.0/UDP 212.227.18.134;branch=z9hG4bKfca6.47b3fa5e1b17ae20c3c2ac4a16c47517.0.
Via: SIP/2.0/UDP 212.227.18.164;branch=z9hG4bKfca6.bfafd08d8fb4802c8a47a88a5541c482.0.
Via: SIP/2.0/UDP 212.227.18.199;branch=z9hG4bKfca6.b20ae9ecb96fb2df8e483de468694f06.0.
Via: SIP/2.0/UDP 195.71.47.156;branch=z9hG4bKfca6.9e613622.0.
Via: SIP/2.0/UDP 195.71.188.6:5060;branch=z9hG4bKi55i2u20787h3l94d7f1.1.
From: +49151XXXXXXXX <sip:+49151XXXXXXXX@1und1-3.sip.mgc.voip.telefonica.de:5060;user=phone>;tag=1358221950.
To: +499133XXXXXXX <sip:+499133XXXXXXX@195.71.188.38:5060;user=phone>.
Call-ID: 7748b0ca-15396ef3-67762b52-d227@1und1-3.sip.mgc.voip.telefonica.de.
CSeq: 1 INVITE.
Max-Forwards: 22.
Supported: timer.
Session-Expires: 1800.
Min-SE: 1800.
Contact: +49151XXXXXXXX <sip:+49151XXXXXXXX@195.71.188.6:5060;transport=udp>.
Allow: INVITE,ACK,PRACK,SUBSCRIBE,BYE,CANCEL,NOTIFY,INFO,REFER,UPDATE.
Content-Type: application/sdp.
Content-Length: 683.
.
v=0.
o=- 690489578 0 IN IP4 62.52.146.44.
s=Cisco SDP 0.
c=IN IP4 62.52.146.44.
t=0 0.
m=audio 31642 RTP/AVP 8 0 18 101 102 103 104 105 4 106 3 107 108 109 125 99 100.
a=rtpmap:101 G729a/8000.
a=rtpmap:102 G726-16/8000.
a=rtpmap:103 G726-24/8000.
a=rtpmap:104 G726-32/8000.
a=rtpmap:105 G7231-H/8000.
a=rtpmap:106 G7231-L/8000.
a=r

U 2014/11/13 13:30:56.360530 192.168.10.207:5060 -> 212.227.18.134:5060
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP 212.227.18.134;branch=z9hG4bKfca6.47b3fa5e1b17ae20c3c2ac4a16c47517.0;received=212.227.18.134;rport=5060.
Via: SIP/2.0/UDP 212.227.18.164;branch=z9hG4bKfca6.bfafd08d8fb4802c8a47a88a5541c482.0.
Via: SIP/2.0/UDP 212.227.18.199;branch=z9hG4bKfca6.b20ae9ecb96fb2df8e483de468694f06.0.
Via: SIP/2.0/UDP 195.71.47.156;branch=z9hG4bKfca6.9e613622.0.
Via: SIP/2.0/UDP 195.71.188.6:5060;branch=z9hG4bKi55i2u20787h3l94d7f1.1.
Record-Route: <sip:212.227.18.134;lr=on>.
Record-Route: <sip:212.227.18.164;lr=on;did=89a.418e8ff>.
Record-Route: <sip:212.227.18.199;lr=on>.
From: +49151XXXXXXXX <sip:+49151XXXXXXXX@1und1-3.sip.mgc.voip.telefonica.de:5060;user=phone>;tag=1358221950.
To: +499133XXXXXXX <sip:+499133XXXXXXX@195.71.188.38:5060;user=phone>.
Call-ID: 7748b0ca-15396ef3-67762b52-d227@1und1-3.sip.mgc.voip.telefonica.de.
CSeq: 1 INVITE.
Server: Asterisk PBX 11.6-cert5.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
Supported: replaces, timer.
Session-Expires: 1800;refresher=uas.
Contact: <sip:499133XXXXXXX@93.XXX.XXX:5060>.
Content-Length: 0.
.


U 2014/11/13 13:30:56.362112 192.168.10.207:5060 -> 212.227.18.134:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 212.227.18.134;branch=z9hG4bKfca6.47b3fa5e1b17ae20c3c2ac4a16c47517.0;received=212.227.18.134;rport=5060.
Via: SIP/2.0/UDP 212.227.18.164;branch=z9hG4bKfca6.bfafd08d8fb4802c8a47a88a5541c482.0.
Via: SIP/2.0/UDP 212.227.18.199;branch=z9hG4bKfca6.b20ae9ecb96fb2df8e483de468694f06.0.
Via: SIP/2.0/UDP 195.71.47.156;branch=z9hG4bKfca6.9e613622.0.
Via: SIP/2.0/UDP 195.71.188.6:5060;branch=z9hG4bKi55i2u20787h3l94d7f1.1.
Record-Route: <sip:212.227.18.134;lr=on>.
Record-Route: <sip:212.227.18.164;lr=on;did=89a.418e8ff>.
Record-Route: <sip:212.227.18.199;lr=on>.
From: +49151XXXXXXXX <sip:+49151XXXXXXXX@1und1-3.sip.mgc.voip.telefonica.de:5060;user=phone>;tag=1358221950.
To: +499133XXXXXXX <sip:+499133XXXXXXX@195.71.188.38:5060;user=phone>;tag=as70d6640a.
Call-ID: 7748b0ca-15396ef3-67762b52-d227@1und1-3.sip.mgc.voip.telefonica.de.
CSeq: 1 INVITE.
Server: Asterisk PBX 11.6-cert5.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
Supported: replaces, timer.
Session-Expires: 1800;refresher=uas.
Contact: <sip:499133XXXXXXX@93.206.XXX.XXX:5060>.
Content-Type: application/sdp.
Require: timer.
Content-Length: 289.
.
v=0.
o=root 1817121162 1817121162 IN IP4 93.206.XXX.XXX.
s=Asterisk PBX 11.6-cert5.
c=IN IP4 93.206.XXX.XXX.
t=0 0.
m=audio 5942 RTP/AVP 3 0 8 101.
a=rtpmap:3 GSM/8000.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.
a=sendrecv.


U 2014/11/13 13:30:56.448394 212.227.18.134:5060 -> 192.168.10.207:5060
ACK sip:499133XXXXXXX@93.206.XXX.XXX:5060 SIP/2.0.
Record-Route: <sip:212.227.18.134;lr=on>.
Record-Route: <sip:212.227.18.164;lr=on>.
Record-Route: <sip:212.227.18.199;lr=on>.
Via: SIP/2.0/UDP 212.227.18.134;branch=z9hG4bKfca6.c7b5fe1eda0e47526a8756d1f5d18e56.0.
Via: SIP/2.0/UDP 212.227.18.164;branch=z9hG4bKfca6.7a898ec5604f9c87e299d0e7d300390e.0.
Via: SIP/2.0/UDP 212.227.18.199;branch=z9hG4bKfca6.0eaccc6301a9fe0c432ac720d3ede58a.0.
Via: SIP/2.0/UDP 195.71.188.6:5060;branch=z9hG4bKrldl3e309o70vn5vv6q1.1.
From: +49151XXXXXXXX <sip:+49151XXXXXXXX@1und1-3.sip.mgc.voip.telefonica.de:5060;user=phone>;tag=1358221950.
To: +499133XXXXXXX <sip:+499133XXXXXXX@195.71.188.38:5060;user=phone>;tag=as70d6640a.
Call-ID: 7748b0ca-15396ef3-67762b52-d227@1und1-3.sip.mgc.voip.telefonica.de.
CSeq: 1 ACK.
Max-Forwards: 23.
Content-Length: 0.
.


U 2014/11/13 13:30:57.222187 192.168.10.207:5060 -> 212.227.18.134:5060
BYE sip:+49151XXXXXXXX@195.71.188.6:5060;transport=udp SIP/2.0.
Via: SIP/2.0/UDP 93.206.XXX.XXX:5060;branch=z9hG4bK4832cae8;rport.
Route: <sip:212.227.18.134;lr=on>,<sip:212.227.18.164;lr=on;did=89a.418e8ff>,<sip:212.227.18.199;lr=on>.
Max-Forwards: 70.
From: +499133XXXXXXX <sip:+499133XXXXXXX@195.71.188.38:5060;user=phone>;tag=as70d6640a.
To: +49151XXXXXXXX <sip:+49151XXXXXXXX@1und1-3.sip.mgc.voip.telefonica.de:5060;user=phone>;tag=1358221950.
Call-ID: 7748b0ca-15396ef3-67762b52-d227@1und1-3.sip.mgc.voip.telefonica.de.
CSeq: 102 BYE.
User-Agent: Asterisk PBX 11.6-cert5.
X-Asterisk-HangupCause: User alerting, no answer.
X-Asterisk-HangupCauseCode: 19.
Content-Length: 0.
.


U 2014/11/13 13:30:57.298224 212.227.18.134:5060 -> 192.168.10.207:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 93.206.XXX.XXX:5060;branch=z9hG4bK4832cae8;rport=5060.
From: +499133XXXXXXX <sip:+499133XXXXXXX@195.71.188.38:5060;user=phone>;tag=as70d6640a.
To: +49151XXXXXXXX <sip:+49151XXXXXXXX@1und1-3.sip.mgc.voip.telefonica.de:5060;user=phone>;tag=1358221950.
Call-ID: 7748b0ca-15396ef3-67762b52-d227@1und1-3.sip.mgc.voip.telefonica.de.
CSeq: 102 BYE.
Record-Route: <sip:212.227.18.199;lr=on>.
Record-Route: <sip:212.227.18.164;lr=on>.
Record-Route: <sip:212.227.18.134;lr=on>.
Content-Length: 0.

Wenn ich jede einzelne IP des 1und1 Clusters als Amt anlegen soll, dann kommen da schnell sehr sehr viele Einträge zusammen…

12

re,

im Normalfall sollten die Anrufe vom Provider die gleiche SIP Domain aufweisen, auch wenn der HOST/IP eine andere ist. Dann müssten diese auch über das gleiche Account laufen können.
Kannst du mal einen Dump machen, wenn der Anruf klappt und einen wenn der Anruf nicht klappt!?
Mal abwarten was der Dump diesbezüglich zeigt.

Grüße
Markus

13

Hier ein Dump wenn der Anruf in den no-auth Context geschickt wird:





U 2014/11/13 14:26:49.285039 212.227.18.137:5060 -> 192.168.10.207:5060
INVITE sip:499133XXXXXXX@93.206.XXX.XX:5060 SIP/2.0.
Record-Route: <sip:212.227.18.137;lr=on>.
Record-Route: <sip:212.227.18.227;lr=on;did=714.c83fb534>.
Record-Route: <sip:212.227.18.138;lr=on>.
Via: SIP/2.0/UDP 212.227.18.137;branch=z9hG4bKe448.bc45315bdd452ce106ede987b50148c4.0.
Via: SIP/2.0/UDP 212.227.18.227;branch=z9hG4bKe448.c6dbe5d3847e84d918b8077af28293df.0.
Via: SIP/2.0/UDP 212.227.18.138;branch=z9hG4bKe448.c7afcc54f4a1d1fefc3e2e52f953dd5f.0.
Via: SIP/2.0/UDP 195.71.47.156;branch=z9hG4bKe448.f81dbb73.0.
Via: SIP/2.0/UDP 195.71.188.9:5060;branch=z9hG4bKv8coc8000gcgsd91e7t0.1.
From: +49151XXXXXXXX <sip:+49151XXXXXXXX@1und1-3.sip.mgc.voip.telefonica.de:5060;user=phone>;tag=369228471.
To: +499133XXXXXXX <sip:+499133XXXXXXX@195.71.188.41:5060;user=phone>.
Call-ID: 5a59cf38-469d0b5e-75545de1-6217@1und1-3.sip.mgc.voip.telefonica.de.
CSeq: 1 INVITE.
Max-Forwards: 22.
Supported: timer.
Session-Expires: 1800.
Min-SE: 1800.
Contact: +49151XXXXXXXX <sip:+49151XXXXXXXX@195.71.188.9:5060;transport=udp>.
Allow: INVITE,ACK,PRACK,SUBSCRIBE,BYE,CANCEL,NOTIFY,INFO,REFER,UPDATE.
Content-Type: application/sdp.
Content-Length: 686.
.
v=0.
o=- 4187182792 0 IN IP4 62.52.146.188.
s=Cisco SDP 0.
c=IN IP4 62.52.146.188.
t=0 0.
m=audio 31618 RTP/AVP 8 0 18 101 102 103 104 105 4 106 3 107 108 109 125 99 100.
a=rtpmap:101 G729a/8000.
a=rtpmap:102 G726-16/8000.
a=rtpmap:103 G726-24/8000.
a=rtpmap:104 G726-32/8000.
a=rtpmap:105 G7231-H/8000.
a=rtpmap:106 G7231-L/8000.
a

U 2014/11/13 14:26:49.287600 192.168.10.207:5060 -> 212.227.18.137:5060
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP 212.227.18.137;branch=z9hG4bKe448.bc45315bdd452ce106ede987b50148c4.0;received=212.227.18.137;rport=5060.
Via: SIP/2.0/UDP 212.227.18.227;branch=z9hG4bKe448.c6dbe5d3847e84d918b8077af28293df.0.
Via: SIP/2.0/UDP 212.227.18.138;branch=z9hG4bKe448.c7afcc54f4a1d1fefc3e2e52f953dd5f.0.
Via: SIP/2.0/UDP 195.71.47.156;branch=z9hG4bKe448.f81dbb73.0.
Via: SIP/2.0/UDP 195.71.188.9:5060;branch=z9hG4bKv8coc8000gcgsd91e7t0.1.
Record-Route: <sip:212.227.18.137;lr=on>.
Record-Route: <sip:212.227.18.227;lr=on;did=714.c83fb534>.
Record-Route: <sip:212.227.18.138;lr=on>.
From: +49151XXXXXXXX <sip:+49151XXXXXXXX@1und1-3.sip.mgc.voip.telefonica.de:5060;user=phone>;tag=369228471.
To: +499133XXXXXXX <sip:+499133XXXXXXX@195.71.188.41:5060;user=phone>.
Call-ID: 5a59cf38-469d0b5e-75545de1-6217@1und1-3.sip.mgc.voip.telefonica.de.
CSeq: 1 INVITE.
Server: Asterisk PBX 11.6-cert5.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
Supported: replaces, timer.
Session-Expires: 1800;refresher=uas.
Contact: <sip:499133XXXXXXX@93.206.XXX.XX:5060>.
Content-Length: 0.
.


U 2014/11/13 14:26:49.289295 192.168.10.207:5060 -> 212.227.18.137:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 212.227.18.137;branch=z9hG4bKe448.bc45315bdd452ce106ede987b50148c4.0;received=212.227.18.137;rport=5060.
Via: SIP/2.0/UDP 212.227.18.227;branch=z9hG4bKe448.c6dbe5d3847e84d918b8077af28293df.0.
Via: SIP/2.0/UDP 212.227.18.138;branch=z9hG4bKe448.c7afcc54f4a1d1fefc3e2e52f953dd5f.0.
Via: SIP/2.0/UDP 195.71.47.156;branch=z9hG4bKe448.f81dbb73.0.
Via: SIP/2.0/UDP 195.71.188.9:5060;branch=z9hG4bKv8coc8000gcgsd91e7t0.1.
Record-Route: <sip:212.227.18.137;lr=on>.
Record-Route: <sip:212.227.18.227;lr=on;did=714.c83fb534>.
Record-Route: <sip:212.227.18.138;lr=on>.
From: +49151XXXXXXXX <sip:+49151XXXXXXXX@1und1-3.sip.mgc.voip.telefonica.de:5060;user=phone>;tag=369228471.
To: +499133XXXXXXX <sip:+499133XXXXXXX@195.71.188.41:5060;user=phone>;tag=as3d7f7e56.
Call-ID: 5a59cf38-469d0b5e-75545de1-6217@1und1-3.sip.mgc.voip.telefonica.de.
CSeq: 1 INVITE.
Server: Asterisk PBX 11.6-cert5.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
Supported: replaces, timer.
Session-Expires: 1800;refresher=uas.
Contact: <sip:499133XXXXXXX@93.206.XXX.XX:5060>.
Content-Type: application/sdp.
Require: timer.
Content-Length: 287.
.
v=0.
o=root 2067296412 2067296412 IN IP4 93.206.XXX.XX.
s=Asterisk PBX 11.6-cert5.
c=IN IP4 93.206.XXX.XX.
t=0 0.
m=audio 6554 RTP/AVP 3 0 8 101.
a=rtpmap:3 GSM/8000.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.
a=sendrecv.


U 2014/11/13 14:26:49.385338 212.227.18.137:5060 -> 192.168.10.207:5060
ACK sip:499133XXXXXXX@93.206.XXX.XX:5060 SIP/2.0.
Record-Route: <sip:212.227.18.137;lr=on>.
Record-Route: <sip:212.227.18.227;lr=on>.
Record-Route: <sip:212.227.18.138;lr=on>.
Via: SIP/2.0/UDP 212.227.18.137;branch=z9hG4bKe448.2f1324bdb5033213d73517fc2c69adf8.0.
Via: SIP/2.0/UDP 212.227.18.227;branch=z9hG4bKe448.4b428be335e08257066a8f256ac14f64.0.
Via: SIP/2.0/UDP 212.227.18.138;branch=z9hG4bKe448.5cc83e9837cbc3c1073d72edaebdbe03.0.
Via: SIP/2.0/UDP 195.71.188.9:5060;branch=z9hG4bK59kf3b3010n1p5p742b0.1.
From: +49151XXXXXXXX <sip:+49151XXXXXXXX@1und1-3.sip.mgc.voip.telefonica.de:5060;user=phone>;tag=369228471.
To: +499133XXXXXXX <sip:+499133XXXXXXX@195.71.188.41:5060;user=phone>;tag=as3d7f7e56.
Call-ID: 5a59cf38-469d0b5e-75545de1-6217@1und1-3.sip.mgc.voip.telefonica.de.
CSeq: 1 ACK.
Max-Forwards: 23.
Content-Length: 0.
.


U 2014/11/13 14:26:50.205560 192.168.10.207:5060 -> 212.227.18.137:5060
BYE sip:+49151XXXXXXXX@195.71.188.9:5060;transport=udp SIP/2.0.
Via: SIP/2.0/UDP 93.206.XXX.XX:5060;branch=z9hG4bK4f9f69a1;rport.
Route: <sip:212.227.18.137;lr=on>,<sip:212.227.18.227;lr=on;did=714.c83fb534>,<sip:212.227.18.138;lr=on>.
Max-Forwards: 70.
From: +499133XXXXXXX <sip:+499133XXXXXXX@195.71.188.41:5060;user=phone>;tag=as3d7f7e56.
To: +49151XXXXXXXX <sip:+49151XXXXXXXX@1und1-3.sip.mgc.voip.telefonica.de:5060;user=phone>;tag=369228471.
Call-ID: 5a59cf38-469d0b5e-75545de1-6217@1und1-3.sip.mgc.voip.telefonica.de.
CSeq: 102 BYE.
User-Agent: Asterisk PBX 11.6-cert5.
X-Asterisk-HangupCause: User alerting, no answer.
X-Asterisk-HangupCauseCode: 19.
Content-Length: 0.
.


U 2014/11/13 14:26:50.285901 212.227.18.137:5060 -> 192.168.10.207:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 93.206.XXX.XX:5060;branch=z9hG4bK4f9f69a1;rport=5060.
From: +499133XXXXXXX <sip:+499133XXXXXXX@195.71.188.41:5060;user=phone>;tag=as3d7f7e56.
To: +49151XXXXXXXX <sip:+49151XXXXXXXX@1und1-3.sip.mgc.voip.telefonica.de:5060;user=phone>;tag=369228471.
Call-ID: 5a59cf38-469d0b5e-75545de1-6217@1und1-3.sip.mgc.voip.telefonica.de.
CSeq: 102 BYE.
Record-Route: <sip:212.227.18.138;lr=on>.
Record-Route: <sip:212.227.18.227;lr=on>.
Record-Route: <sip:212.227.18.137;lr=on>.
Content-Length: 0.

Und hier wenn es mal funktioniert:

U 2014/11/13 15:19:11.011771 212.227.67.132:5060 -> 192.168.10.207:5060
INVITE sip:499133XXXXXXX@93.206.XXX.XX:5060 SIP/2.0.
Record-Route: <sip:212.227.67.132;lr=on>.
Record-Route: <sip:212.227.67.226;lr=on;did=a23.5a8c4931>.
Record-Route: <sip:212.227.67.139;lr=on>.
Via: SIP/2.0/UDP 212.227.67.132;branch=z9hG4bK1633.528d52282a20af3c997d31d787fa96b9.0.
Via: SIP/2.0/UDP 212.227.67.226;branch=z9hG4bK1633.214576c83f8c08d15930fee8f8071341.0.
Via: SIP/2.0/UDP 212.227.67.139;branch=z9hG4bK1633.d713a4e7fd4f5173b2953aa59652aceb.0.
Via: SIP/2.0/UDP 217.188.44.83;branch=z9hG4bK1633.4cba65f.0.
Via: SIP/2.0/UDP 195.71.188.6:5060;branch=z9hG4bKktt7lu00e0i04optp2v1.1.
From: +49151XXXXXXXX <sip:+49151XXXXXXXX@1und1-5.sip.mgc.voip.telefonica.de:5060;user=phone>;tag=1234948348.
To: +499133XXXXXXX <sip:+499133XXXXXXX@195.71.188.38:5060;user=phone>.
Call-ID: 48f95e27-4b9b2469-7a5b7912-dd8a@1und1-5.sip.mgc.voip.telefonica.de.
CSeq: 1 INVITE.
Max-Forwards: 22.
Supported: timer.
Session-Expires: 1800.
Min-SE: 1800.
Contact: +49151XXXXXXXX <sip:+49151XXXXXXXX@195.71.188.6:5060;transport=udp>.
Allow: INVITE,ACK,PRACK,SUBSCRIBE,BYE,CANCEL,NOTIFY,INFO,REFER,UPDATE.
Content-Type: application/sdp.
Content-Length: 380.
.
v=0.
o=- 1767769624 1415888349 IN IP4 62.52.146.44.
s=-.
c=IN IP4 62.52.146.44.
t=0 0.
m=audio 31858 RTP/AVP 8 0 18 104 107 125 100 101.
a=rtpmap:104 G726-32/8000/1.
a=rtpmap:107 G729/8000/1.
a=rtpmap:125 CLEARMODE/8000/1.
a=fmtp:107 annexb=yes.
a=fmtp:18 annexb=no.
a=ptime:20.
a=rtpmap:100 X-NSE/8000/1.
a=fmtp:100 192-194.
a=rtpm

U 2014/11/13 15:19:11.079286 192.168.10.207:5060 -> 212.227.67.132:5060
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP 212.227.67.132;branch=z9hG4bK1633.528d52282a20af3c997d31d787fa96b9.0;received=212.227.67.132;rport=5060.
Via: SIP/2.0/UDP 212.227.67.226;branch=z9hG4bK1633.214576c83f8c08d15930fee8f8071341.0.
Via: SIP/2.0/UDP 212.227.67.139;branch=z9hG4bK1633.d713a4e7fd4f5173b2953aa59652aceb.0.
Via: SIP/2.0/UDP 217.188.44.83;branch=z9hG4bK1633.4cba65f.0.
Via: SIP/2.0/UDP 195.71.188.6:5060;branch=z9hG4bKktt7lu00e0i04optp2v1.1.
Record-Route: <sip:212.227.67.132;lr=on>.
Record-Route: <sip:212.227.67.226;lr=on;did=a23.5a8c4931>.
Record-Route: <sip:212.227.67.139;lr=on>.
From: +49151XXXXXXXX <sip:+49151XXXXXXXX@1und1-5.sip.mgc.voip.telefonica.de:5060;user=phone>;tag=1234948348.
To: +499133XXXXXXX <sip:+499133XXXXXXX@195.71.188.38:5060;user=phone>.
Call-ID: 48f95e27-4b9b2469-7a5b7912-dd8a@1und1-5.sip.mgc.voip.telefonica.de.
CSeq: 1 INVITE.
Server: Asterisk PBX 11.6-cert5.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
Supported: replaces, timer.
Session-Expires: 1800;refresher=uas.
Contact: <sip:499133XXXXXXX@93.206.XXX.XX:5060>.
Content-Length: 0.
.


U 2014/11/13 15:19:11.351149 192.168.10.207:5060 -> 192.168.10.20:5060
INVITE sip:creadoo1@192.168.10.20:5060 SIP/2.0.
Via: SIP/2.0/UDP 192.168.10.207:5060;branch=z9hG4bK5dffa746;rport.
Max-Forwards: 70.
From: "Sven" <sip:00151XXXXXXXX@192.168.10.207>;tag=as5e0d5c62.
To: <sip:creadoo1@192.168.10.20:5060>.
Contact: <sip:00151XXXXXXXX@192.168.10.207:5060>.
Call-ID: 5800249522872e3c499263d6200e98b1@192.168.10.207:5060.
CSeq: 102 INVITE.
User-Agent: Asterisk PBX 11.6-cert5.
Date: Thu, 13 Nov 2014 14:19:11 GMT.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
Supported: replaces, timer.
Alert-Info: <http://www.notused.de>;info=alert-external;x-line-id=0.
Content-Type: application/sdp.
Content-Length: 455.
.
v=0.
o=root 1379483451 1379483451 IN IP4 192.168.10.207.
s=Asterisk PBX 11.6-cert5.
c=IN IP4 192.168.10.207.
b=CT:384.
t=0 0.
m=audio 30758 RTP/AVP 0 8 3 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:3 GSM/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.
a=sendrecv.
m=video 9618 RTP/AVP 99.
a=rtpmap:99 H264/90000.
a=fmtp:99 redundant-pic-cap=0;parameter-add=0;packetization-mode=0;level-asymmetry-allowed=0.
a=sendrecv.


U 2014/11/13 15:19:11.450989 192.168.10.207:5060 -> 192.168.10.20:5060
INVITE sip:creadoo1@192.168.10.20:5060 SIP/2.0.
Via: SIP/2.0/UDP 192.168.10.207:5060;branch=z9hG4bK5dffa746;rport.
Max-Forwards: 70.
From: "Sven" <sip:00151XXXXXXXX@192.168.10.207>;tag=as5e0d5c62.
To: <sip:creadoo1@192.168.10.20:5060>.
Contact: <sip:00151XXXXXXXX@192.168.10.207:5060>.
Call-ID: 5800249522872e3c499263d6200e98b1@192.168.10.207:5060.
CSeq: 102 INVITE.
User-Agent: Asterisk PBX 11.6-cert5.
Date: Thu, 13 Nov 2014 14:19:11 GMT.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
Supported: replaces, timer.
Alert-Info: <http://www.notused.de>;info=alert-external;x-line-id=0.
Content-Type: application/sdp.
Content-Length: 455.
.
v=0.
o=root 1379483451 1379483451 IN IP4 192.168.10.207.
s=Asterisk PBX 11.6-cert5.
c=IN IP4 192.168.10.207.
b=CT:384.
t=0 0.
m=audio 30758 RTP/AVP 0 8 3 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:3 GSM/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.
a=sendrecv.
m=video 9618 RTP/AVP 99.
a=rtpmap:99 H264/90000.
a=fmtp:99 redundant-pic-cap=0;parameter-add=0;packetization-mode=0;level-asymmetry-allowed=0.
a=sendrecv.


U 2014/11/13 15:19:11.497706 192.168.10.20:5060 -> 192.168.10.207:5060
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP 192.168.10.207:5060;branch=z9hG4bK5dffa746;rport=5060.
From: "Sven" <sip:00151XXXXXXXX@192.168.10.207>;tag=as5e0d5c62.
To: <sip:creadoo1@192.168.10.20:5060>;tag=3024718246.
Call-ID: 5800249522872e3c499263d6200e98b1@192.168.10.207:5060.
CSeq: 102 INVITE.
Contact: <sip:creadoo1@192.168.10.20:5060>.
Content-Length: 0.
.


U 2014/11/13 15:19:11.525600 192.168.10.20:5060 -> 192.168.10.207:5060
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP 192.168.10.207:5060;branch=z9hG4bK5dffa746;rport=5060.
From: "Sven" <sip:00151XXXXXXXX@192.168.10.207>;tag=as5e0d5c62.
To: <sip:creadoo1@192.168.10.20:5060>;tag=3024718246.
Call-ID: 5800249522872e3c499263d6200e98b1@192.168.10.207:5060.
CSeq: 102 INVITE.
Contact: <sip:creadoo1@192.168.10.20:5060>.
Content-Length: 0.
.


U 2014/11/13 15:19:12.704543 192.168.10.20:5060 -> 192.168.10.207:5060
SIP/2.0 180 Ringing.
Via: SIP/2.0/UDP 192.168.10.207:5060;branch=z9hG4bK5dffa746;rport=5060.
From: "Sven" <sip:00151XXXXXXXX@192.168.10.207>;tag=as5e0d5c62.
To: <sip:creadoo1@192.168.10.20:5060>;tag=3024718246.
Call-ID: 5800249522872e3c499263d6200e98b1@192.168.10.207:5060.
CSeq: 102 INVITE.
Contact: <sip:creadoo1@192.168.10.20:5060>.
Allow-Events: message-summary, refer, talk.
Content-Length: 0.
.


U 2014/11/13 15:19:12.705775 192.168.10.207:5060 -> 212.227.67.132:5060
SIP/2.0 180 Ringing.
Via: SIP/2.0/UDP 212.227.67.132;branch=z9hG4bK1633.528d52282a20af3c997d31d787fa96b9.0;received=212.227.67.132;rport=5060.
Via: SIP/2.0/UDP 212.227.67.226;branch=z9hG4bK1633.214576c83f8c08d15930fee8f8071341.0.
Via: SIP/2.0/UDP 212.227.67.139;branch=z9hG4bK1633.d713a4e7fd4f5173b2953aa59652aceb.0.
Via: SIP/2.0/UDP 217.188.44.83;branch=z9hG4bK1633.4cba65f.0.
Via: SIP/2.0/UDP 195.71.188.6:5060;branch=z9hG4bKktt7lu00e0i04optp2v1.1.
Record-Route: <sip:212.227.67.132;lr=on>.
Record-Route: <sip:212.227.67.226;lr=on;did=a23.5a8c4931>.
Record-Route: <sip:212.227.67.139;lr=on>.
From: +49151XXXXXXXX <sip:+49151XXXXXXXX@1und1-5.sip.mgc.voip.telefonica.de:5060;user=phone>;tag=1234948348.
To: +499133XXXXXXX <sip:+499133XXXXXXX@195.71.188.38:5060;user=phone>;tag=as429cded6.
Call-ID: 48f95e27-4b9b2469-7a5b7912-dd8a@1und1-5.sip.mgc.voip.telefonica.de.
CSeq: 1 INVITE.
Server: Asterisk PBX 11.6-cert5.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
Supported: replaces, timer.
Session-Expires: 1800;refresher=uas.
Contact: <sip:499133XXXXXXX@93.206.XXX.XX:5060>.
Content-Length: 0.
14

Re,

das mit den mehreren Accounts können wir glaub ich aufgrund der Vielzahl nicht wirklich umsetzen.
Eine Alternative wäre, dass wir unter Module->Skripte das no-auth-in-Skript anpassen, so dass wir hier den eingehenden SIP Header zerlegen und entsprechend validieren. So ähnlich als wie wenn du beim Amt “Ext aus Header” gewählt hättest. Anschließend dann auf den eingehenden Kontext, des Amtes weiterleiten - mdc_incoming-X, welches du schon hast, so dass die eingehenden Regeln greifen. Dann bleibt auch soweit alles über die GUI verwaltbar.
Ich nehme hierzu auf jeden Fall ein Ticket auf.
Wenn es sehr dringend ist, kannst du dich auch an unseren Support direkt wenden.

Grüße
Markus

15

Hallo Markus,

vielen Dank für das Ticket und Deine Hilfe.
Ich habe mir in der Zwischenzeit damit beholfen, dass ich den no-auth Context auf mdc_incoming-X gesetzt habe, allowguest=yes und in der Firewall nur die entsprechenden IP-Adressen von sip.1und1 durchlasse.
Nachteil ist natürlich, dass ich keine vernünftigen Rufregeln so hinbekomme.
Aber bis zur Scriptvariante sollte es reichen. Zumindest sind wir erreichbar :slight_smile:
Eigentlich müsste es ja alle betreffen, die einen 1und1 Anschluss haben?!

Liebe Grüße

Sven

16

Hallo Sven,

kenn jetzt die Tarife/Produkte von 1und1 nicht, aber wenn sie es so ähnlich machen wie andere Anbieter, dann kann schon sein, dass sie für jede Produktvariante eine anderen Weg gehen.
Nichtsdestotrotz wir werden in den nächsten Versionen eine komfortable Lösung hierzu anbietern.

Grüße
Markus

17

Hallo,

gibt es schon Fortschritte bei 1und1?

Ich bekomme ebenfalls bei eingehenden Anrufen ein Playing ‘beeperr.slin’ (language ‘en’) vorgespielt und mit 0 Sekunden wird der Anruf aufgelegt trotz Regeln für den Eingang.

Ich nutze seit Jahren eine 3CX hinter NAT und das funktioniert reibungslos (mit1&1). Die Registrierung läuft zu 1und1 auf udp5060 wie es sein soll zur 212.227.0.0/16.
Allerdings kommt bei eingehendem Anruf immer was zurück von der Telefonica 62.52.0.0/14 auf 5060 zur 3CX.

Kann es event. an diesem anderen Netz liegen, daß es mit den eingehenden Anrufen auf der MD nicht klappt, da meldet sich nämlich Telefonica nicht?

Jemand eine Idee.

Gruß Oliver

18

Ich habe mir damit beholfen, dass ich per Firewall nur aus dem IP-Bereich von 1&1 / Telefonica Anfragen auf Port 5060 durchlasse.
In der MobyDick dann den “no-auth” context auf mdc_incoming-X gesetzt und allowguest=yes.
So werden die eingehenden Anrufe wenigstens nicht verworfen.
Eine saubere Lösung ist es aber so nicht wirklich.
Aber es funktioniert.
Die Funktion allowguest=yes bitte nur verwenden, wenn Du wirklich die IP Adressen bestimmen kannst, die auf die Anlage weitergeleitet werden. Sonst hast Du ein offenes Relay.

19

Ja Danke, rein lasse ich durch die FW nur die Telefonica/1und1 Netze.

aber wo stelle ich

" In der MobyDick dann den “no-auth” context auf mdc_incoming-X gesetzt und allowguest=yes."

ein? In den Systemeinstellungen finde ich nix, etwa in der sip.conf? Ich dachte, diese wird immer überschrieben.

20

Ja, direkt in der sip.conf.
Damit die Einstellungen nicht überschrieben werden, unter Systemeinstellungen->sys->asterisk->configure->sip den Wert Managed auf 0 setzen.

Der Wert X bei mdc_incoming-X bezieht sich auf einen real existierenden trunk.