Probleme mit 1und1

Mandarine · January 15, 2015, 4:24am

[Jan 15 13:21:52] NOTICE[22067][C-00000000]: chan_sip.c:25452 handle_request_invite: Call from ‘’ (212.227.18.132:5060) to extension ‘4991xxxxxx’ rejected because extension not found in context ‘mdc_incoming-Amt’.

also meine eingehende Regel lautet >>>> Amt
trotzdem nicht erreichbar.
Und wie gestaltet sich das Ganze denn mit mehreren SIP 1und1?

Pandos · January 15, 2015, 4:29am

Bei den Ämtern erkennt MobyDick ja automatisch, ob es ein Multiamt ist, oder nicht.
Nun müsstest Du noch die Regel der eingehenden Rufe anpassen.
Quelle:*
Ziel: Deine Rufnummer im Format: 49XX oder *wenn alle Nummern gelten sollen
Durchwahl: Deine Durchwahl

Mandarine · January 15, 2015, 5:15am

das Ziel hatte ich bereits mit * und Durchwahl >>>>>>>>>> rejected because extension not found in context 'mdc_incoming-Amt

mit Nummer als Ziel das selbe.
Ist denn die Bezeichnung Amt bei eingehende Anrufe jene von mdc_incoming-Amt?

in den Optionen bei account steht bei mir
fromdomain=sip.1und1.de
fromuser=4991xxxxxxxxxxx
insecure=invite
disallow=all
allow=alaw
allow=ulaw
nat=force_rport,comedia

[general]
bindaddr=0.0.0.0
context=mdc_incoming-Amt
notifyringing=yes
port=5060
rtpholdtimeout=600
rtptimeout=60
srvlookup=yes
callevents=yes
allowsubscribe=yes
notifyhold=yes
limitonpeers=yes
callcounter=yes
transport=udp
;encryption=yes
notifycid=ignore-context
qualify=yes
pedantic=no
useclientcode=yes
defaultexpirey=600
tos_sip=cs3
tos_audio=ef
tos_video=af41
tos_text=cs3
dtmfmode=rfc2833
language=de
allowguest=yes
videosupport=yes
canreinvite=no
externhost=xxxxxxxxxxx.changeip.net
localnet=10.66.3.0/255.255.255.0
#include mdc_sip_register.conf
#include mdc_sip_ipdevice.conf
#include mdc_sip_trunk.conf
#include mdc_sip_gw.conf

interne Wählerei geht problemlos, raus auch. Nur rein kein einziges mal.
und icinga meldet sich zu Wort: asterisk SIP SIP CRITICAL MISSING 404 !!!

Pandos · January 15, 2015, 6:25am

Schau mal bitte nach, ob mdc_incoming-Amt identisch ist mit dem context in der mdc_sip_trunk.conf unter der Telefonnummer.
Hey, auch jemand aus dem 091er Bereich

Außerdem könntest Du evtl. mal versuchen in der sip.conf den Befehl auf insecure=port,invite zu setzen.

Mandarine · January 15, 2015, 10:47am

Ja, mit den Werten aus mdc_sip_trunk.conf geht es natürlich, auch wenn mehrere accountsangelegt sind kann man unterscheien und braucht nur einen mdc_incoming-wasauchimmer Eintrag.

Soweit so gut. Komisch ist nur, daß mir als anrufender Teilnehmer 00meineVorwahl49Anrufer angezeigt wird, sodaß ich darauf nicht zurückrufen kann. Sehr seltsam.

Markus · January 15, 2015, 11:20pm

Guten Morgen,

falls du immer noch ein Problem mit den eingehenden Anrufen hast

[Jan 15 13:21:52] NOTICE[22067][C-00000000]: chan_sip.c:25452 handle_request_invite: Call from ‘’ (212.227.18.132:5060) to extension ‘4991xxxxxx’ rejected because extension not found in context ‘mdc_incoming-Amt’.

Die Kontexte welche bei den Accounts verwendet werden, heißen z.B. mdc_incoming-X, dort sind dann auch die eingehenden Regeln hinterlegt. Falls du einen eigenen Kontext mit mdc_incoming-Amt verwendest, musst du entweder ein Skript mit dem Kontext als Namen erzeugen und dort die Anrufe auf einen mdc_incoming-X weiterleiten oder du trägst bei jedem Account, wenn du mehrere Ämter hast (und nicht pro Amt mehrere Accounts) den Kontext des Amtes ein, welches die eingehenden Regeln beinhaltet.

Gruß
Markus

Mandarine · January 15, 2015, 11:44pm

ja, Danke, das mit der mdc_sip_trunk.conf habe ich nun schon verstanden, es klappt natürlich mit den eingehenden Anrufen damit. Für einen Asterisk Neuling ist das alles nicht so leicht.

Verwundert bin ich nur, warum ich als Anrufer erst 00 gefolgt von meiner Vorwahl und dann die Rufnummer des Anrufers angezeigt bekomme. So kann man nicht zurückrufen.

Markus · January 16, 2015, 12:07am

Re,

Hast du beim Amt alle Kennzeichen hinterlegt (LKz, NatVaz, IntVaz, OnKz)? oder nicht. Poste vielleicht mal einen kompletten Auszug der CLI für einen eingehenden Anruf. Dann kann ich dir sagen, ob bei der Normierung der Rufnummer auf das internationale Format bzw. bei der Verwendung der Präfixe etwas nicht klappt.
Verwendest du bei den ausgehenden Regeln einen In-Präfix z.B. 0, dann solltest du damit der Rückruf klappt auch beim Amt (Basisdaten) bei Präfix eing. Nummer eine 0 eintragen.

Gruß
Markus

Mandarine · January 16, 2015, 2:29am

ja, sind hinterlegt LKz, NatVaz, IntVaz, OnKz.

bei dem account, der als Benutzername mit 09 beginnt, klappt die Rufnummernanzeige wunderbar.

Connected to Asterisk 11.6-cert5 currently running on PBX (pid = 7307)
== 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 [09199999999@mdc_incoming-6:1] Set(“SIP/87.234.1.132-00000000”, “MDC_CALLER_NUM_TRUNK=01718888888”) in new stack
– Executing [09199999999@mdc_incoming-6:2] Set(“SIP/87.234.1.132-00000000”, “MDC_CALLEE_NUM_TRUNK=09199999999”) in new stack
– Executing [09199999999@mdc_incoming-6:3] Goto(“SIP/87.234.1.132-00000000”, “mdc_trunk-3,s,1”) in new stack
– Goto (mdc_trunk-3,s,1)
– Executing [s@mdc_trunk-3:1] Verbose(“SIP/87.234.1.132-00000000”, “1,callee number: 09199999999 caller number: 01718888888”) in new stack
callee number: 09199999999 caller number: 01718888888
– Executing [s@mdc_trunk-3:2] Gosub(“SIP/87.234.1.132-00000000”, “sub_nat2int,s,1(MDC_CALLER_NUM_INTERNAT,01718888888,00,49,0,9192)”) in new stack
– Executing [s@sub_nat2int:1] Verbose(“SIP/87.234.1.132-00000000”, “1,sub_nat2int:: variable: MDC_CALLER_NUM_INTERNAT - number: 01718888888 - intVAZ: 00 - lkz: 49 - natVAZ: 0 - onKz: 9192”) in new stack
sub_nat2int:: variable: MDC_CALLER_NUM_INTERNAT - number: 01718888888 - intVAZ: 00 - lkz: 49 - natVAZ: 0 - onKz: 9192
– Executing [s@sub_nat2int:2] GotoIf(“SIP/87.234.1.132-00000000”, “0?s-emergency,1”) in new stack
– Executing [s@sub_nat2int:3] GotoIf(“SIP/87.234.1.132-00000000”, “0?s-int,1”) in new stack
– Executing [s@sub_nat2int:4] GotoIf(“SIP/87.234.1.132-00000000”, “0?s-int,1”) in new stack
– Executing [s@sub_nat2int:5] GotoIf(“SIP/87.234.1.132-00000000”, " 0?s-convert,1") in new stack
– Executing [s@sub_nat2int:6] GotoIf(“SIP/87.234.1.132-00000000”, “0?s-int,1”) in new stack
– Executing [s@sub_nat2int:7] GotoIf(“SIP/87.234.1.132-00000000”, “0?s-intshort,1”) in new stack
– Executing [s@sub_nat2int:8] GotoIf(“SIP/87.234.1.132-00000000”, “0?s-natshort,1”) in new stack
– Executing [s@sub_nat2int:9] GotoIf(“SIP/87.234.1.132-00000000”, “1?s-nat,1”) in new stack
– Goto (sub_nat2int,s-nat,1)
– Executing [s-nat@sub_nat2int:1] Verbose(“SIP/87.234.1.132-00000000”, “1,national”) in new stack
national
– Executing [s-nat@sub_nat2int:2] Set(“SIP/87.234.1.132-00000000”, “MDC_CALLER_NUM_INTERNAT=00491718888888”) in new stack
– Executing [s-nat@sub_nat2int:3] Return(“SIP/87.234.1.132-00000000”, “”) in new stack
– Executing [s@mdc_trunk-3:3] Set(“SIP/87.234.1.132-00000000”, “CALLERID(num)=00491718888888”) in new stack
– Executing [s@mdc_trunk-3:4] Gosub(“SIP/87.234.1.132-00000000”, “sub_int2nat,s,1(MDC_CALLER_NUM_NAT,00491718888888,00,49,0,9192)”) in new stack
– Executing [s@sub_int2nat:1] Verbose(“SIP/87.234.1.132-00000000”, “1,sub_int2nat:: variable: MDC_CALLER_NUM_NAT - exten: 00491718888888 - intVAZ: 00 - lkz: 49 - natVAZ: 0 - onKz: 9192”) in new stack
sub_int2nat:: variable: MDC_CALLER_NUM_NAT - exten: 00491718888888 - intVAZ: 00 - lkz: 49 - natVAZ: 0 - onKz: 9192
– Executing [s@sub_int2nat:2] GotoIf(“SIP/87.234.1.132-00000000”, “0?s-emergency,1”) in new stack
– Executing [s@sub_int2nat:3] GotoIf(“SIP/87.234.1.132-00000000”, “0?nat”) in new stack
– Executing [s@sub_int2nat:4] GotoIf(“SIP/87.234.1.132-00000000”, “1?s-internat,1”) in new stack
– Goto (sub_int2nat,s-internat,1)
– Executing [s-internat@sub_int2nat:1] Set(“SIP/87.234.1.132-00000000”, “MDC_CALLER_NUM_NAT=01718888888”) in new stack
– Executing [s-internat@sub_int2nat:2] Return(“SIP/87.234.1.132-00000000”, “”) in new stack
– Executing [s@mdc_trunk-3:5] Set(“SIP/87.234.1.132-00000000”, “CALLERID(num)=01718888888”) in new stack
– Executing [s@mdc_trunk-3:6] UserEvent(“SIP/87.234.1.132-00000000”, “ResolveCallerName,Strategy: default,Outbound: 0,Channel: SIP/87.234.1.132-00000000”) in new stack
– Executing [s@mdc_trunk-3:7] Wait(“SIP/87.234.1.132-00000000”, “0.25”) in new stack
– Executing [s@mdc_trunk-3:8] Verbose(“SIP/87.234.1.132-00000000”, “1,MDC_RESOLVENAME_HITS = 0”) in new stack
MDC_RESOLVENAME_HITS = 0
– Executing [s@mdc_trunk-3:9] Verbose(“SIP/87.234.1.132-00000000”, "1,CALLERID(name) = ") in new stack
CALLERID(name) =
– Executing [s@mdc_trunk-3:10] Set(“SIP/87.234.1.132-00000000”, “MDC_NUMPREFIX_TRUNK=0”) in new stack
– Executing [s@mdc_trunk-3:11] ExecIf(“SIP/87.234.1.132-00000000”, “1?Set(CALLERID(num)=001718888888)”) in new stack
– Executing [s@mdc_trunk-3:12] Goto(“SIP/87.234.1.132-00000000”, “mdc_mapping-3,09199999999,1”) in new stack
– Goto (mdc_mapping-3,09199999999,1)
– Executing [09199999999@mdc_mapping-3:1] Set(“SIP/87.234.1.132-00000000”, “CHANNEL(language)=de”) in new stack
– Executing [09199999999@mdc_mapping-3:2] Verbose(“SIP/87.234.1.132-00000000”, “1,mapping from 09199999999 to 555”) in new stack
mapping from 09199999999 to 555
– Executing [09199999999@mdc_mapping-3:3] Goto(“SIP/87.234.1.132-00000000”, “mdc_external,555,1”) in new stack
– Goto (mdc_external,555,1)
– Executing [555@mdc_external:1] SIPAddHeader(“SIP/87.234.1.132-00000000”, "“Alert-Info:<http://www.notused.de>;info=alert-external;x-line-id=0"”) in new stack
– Executing [555@mdc_external:2] GosubIf(“SIP/87.234.1.132-00000000”, “1?sub_initcall,s,1(ext,555)”) in new stack
– Executing [s@sub_initcall:1] Verbose(“SIP/87.234.1.132-00000000”, “1,sub_initcall descent: ext exten: 555”) in new stack
sub_initcall descent: ext exten: 555
– Executing [s@sub_initcall:2] GosubIf(“SIP/87.234.1.132-00000000”, “1?sub_initloop,s,1”) in new stack
– Executing [s@sub_initloop:1] Verbose(“SIP/87.234.1.132-00000000”, “1,initial loop”) in new stack
initial loop
– Executing [s@sub_initloop:2] Set(“SIP/87.234.1.132-00000000”, “MDC_ALIAS_HOP=0”) in new stack
– Executing [s@sub_initloop:3] Return(“SIP/87.234.1.132-00000000”, “”) in new stack
– Executing [s@sub_initcall:3] Set(“SIP/87.234.1.132-00000000”, “__MDC_TRANSFERBACK_HOP=0”) in new stack
– Executing [s@sub_initcall:4] Set(“SIP/87.234.1.132-00000000”, “__MDC_DIALDESCENT=ext”) in new stack
– Executing [s@sub_initcall:5] Goto(“SIP/87.234.1.132-00000000”, “ext,1”) in new stack
– Goto (sub_initcall,ext,1)
– Executing [ext@sub_initcall:1] Set(“SIP/87.234.1.132-00000000”, “__MDC_DIALCALLERNUMINIT=01718888888”) in new stack
– Executing [ext@sub_initcall:2] Set(“SIP/87.234.1.132-00000000”, “__MDC_DIALCALLEENUMINIT=09199999999”) in new stack
– Executing [ext@sub_initcall:3] Set(“SIP/87.234.1.132-00000000”, “__MDC_DIALCALLEENUMMAP=555”) in new stack
– Executing [ext@sub_initcall:4] Return(“SIP/87.234.1.132-00000000”, “”) in new stack
– Executing [555@mdc_external:3] Goto(“SIP/87.234.1.132-00000000”, “main,555,1”) in new stack
– Goto (main,555,1)
– Executing [555@main:1] Gosub(“SIP/87.234.1.132-00000000”, “sub_defcall,s,1(555)”) in new stack
– Executing [s@sub_defcall:1] Set(“SIP/87.234.1.132-00000000”, “__MDC_ALIAS_HOP=1”) in new stack
– Executing [s@sub_defcall:2] Set(“SIP/87.234.1.132-00000000”, “__MDC_DIALCALLEENUM=555”) in new stack
– Executing [s@sub_defcall:3] Set(“SIP/87.234.1.132-00000000”, “__MDC_DIALCALLERNUM=001718888888”) in new stack
– Executing [s@sub_defcall:4] GotoIf(“SIP/87.234.1.132-00000000”, “1?nozap”) in new stack
– Goto (sub_defcall,s,8)
– Executing [s@sub_defcall:8] Set(“SIP/87.234.1.132-00000000”, “__MDC_DIALCALLERNAME=”) in new stack
– Executing [s@sub_defcall:9] Set(“SIP/87.234.1.132-00000000”, “__MDC_DIALCHANNELNAME=”) in new stack
– Executing [s@sub_defcall:10] Return(“SIP/87.234.1.132-00000000”, “”) in new stack
– Executing [555@main:2] Goto(“SIP/87.234.1.132-00000000”, “mdc_distribute,555,1”) in new stack
– Goto (mdc_distribute,555,1)
– Executing [555@mdc_distribute:1] Gosub(“SIP/87.234.1.132-00000000”, “sub_team,s,1(1,Gräfenberg,555)”) in new stack
– Executing [s@sub_team:1] Verbose(“SIP/87.234.1.132-00000000”, “1,team id: 1 team name: Gräfenberg team realname: own vm: 555”) in new stack
team id: 1 team name: Gräfenberg team realname: own vm: 555
– Executing [s@sub_team:2] Set(“SIP/87.234.1.132-00000000”, “_MDC_CALLEE_TM_ID=1”) in new stack
– Executing [s@sub_team:3] Set(“SIP/87.234.1.132-00000000”, “_MDC_CALLEE_TM_NAME=Gräfenberg”) in new stack
– Executing [s@sub_team:4] Set(“SIP/87.234.1.132-00000000”, “_MDC_CALLEE_TM_REALNAME=”) in new stack
– Executing [s@sub_team:5] Set(“SIP/87.234.1.132-00000000”, “_MDC_CALLEE_VM_OWN=555”) in new stack
– Executing [s@sub_team:6] Return(“SIP/87.234.1.132-00000000”, “”) in new stack
– Executing [555@mdc_distribute:2] Goto(“SIP/87.234.1.132-00000000”, “mdc_team-1,555,1”) in new stack
– Goto (mdc_team-1,555,1)
– Executing [555@mdc_team-1:1] NoOp(“SIP/87.234.1.132-00000000”, “alias-check:: call forwarding from 555 - 0”) in new stack
– Executing [555@mdc_team-1:2] GotoIf(“SIP/87.234.1.132-00000000”, “0?555-uml,1:555-dial,1”) in new stack
– Goto (mdc_team-1,555-dial,1)
– Executing [555-dial@mdc_team-1:1] Set(“SIP/87.234.1.132-00000000”, “__MDC_EXTEN=555”) in new stack
– Executing [555-dial@mdc_team-1:2] Gosub(“SIP/87.234.1.132-00000000”, “sub_prefix-555,ext,1”) in new stack
– Executing [ext@sub_prefix-555:1] Verbose(“SIP/87.234.1.132-00000000”, “1,no action”) in new stack
no action

Mandarine · January 16, 2015, 2:29am

bei dem weiteren account, der als Benutzername und Durchwahl reg mit 49 beginnt, kommt der Unsinn mit der Vorwahl

== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
– Executing [4909177777777@mdc_incoming-6:1] Set(“SIP/mdc_trunk_conf-6-00000002”, “MDC_CALLER_NUM_TRUNK=491718888888”) in new stack
– Executing [4909177777777@mdc_incoming-6:2] Set(“SIP/mdc_trunk_conf-6-00000002”, “MDC_CALLEE_NUM_TRUNK=4909177777777”) in new stack
– Executing [4909177777777@mdc_incoming-6:3] Goto(“SIP/mdc_trunk_conf-6-00000002”, “mdc_trunk-3,s,1”) in new stack
– Goto (mdc_trunk-3,s,1)
– Executing [s@mdc_trunk-3:1] Verbose(“SIP/mdc_trunk_conf-6-00000002”, “1,callee number: 4909177777777 caller number: 491718888888”) in new stack
callee number: 4909177777777 caller number: 491718888888
– Executing [s@mdc_trunk-3:2] Gosub(“SIP/mdc_trunk_conf-6-00000002”, “sub_nat2int,s,1(MDC_CALLER_NUM_INTERNAT,491718888888,00,49,0,91)”) in new stack
– Executing [s@sub_nat2int:1] Verbose(“SIP/mdc_trunk_conf-6-00000002”, “1,sub_nat2int:: variable: MDC_CALLER_NUM_INTERNAT - number: 491718888888 - intVAZ: 00 - lkz: 49 - natVAZ: 0 - onKz: 91”) in new stack
sub_nat2int:: variable: MDC_CALLER_NUM_INTERNAT - number: 491718888888 - intVAZ: 00 - lkz: 49 - natVAZ: 0 - onKz: 91
– Executing [s@sub_nat2int:2] GotoIf(“SIP/mdc_trunk_conf-6-00000002”, “0?s-emergency,1”) in new stack
– Executing [s@sub_nat2int:3] GotoIf(“SIP/mdc_trunk_conf-6-00000002”, “0?s-int,1”) in new stack
– Executing [s@sub_nat2int:4] GotoIf(“SIP/mdc_trunk_conf-6-00000002”, “0?s-int,1”) in new stack
– Executing [s@sub_nat2int:5] GotoIf(“SIP/mdc_trunk_conf-6-00000002”, " 0?s-convert,1") in new stack
– Executing [s@sub_nat2int:6] GotoIf(“SIP/mdc_trunk_conf-6-00000002”, “0?s-int,1”) in new stack
– Executing [s@sub_nat2int:7] GotoIf(“SIP/mdc_trunk_conf-6-00000002”, “0?s-intshort,1”) in new stack
– Executing [s@sub_nat2int:8] GotoIf(“SIP/mdc_trunk_conf-6-00000002”, “0?s-natshort,1”) in new stack
– Executing [s@sub_nat2int:9] GotoIf(“SIP/mdc_trunk_conf-6-00000002”, “0?s-nat,1”) in new stack
– Executing [s@sub_nat2int:10] Goto(“SIP/mdc_trunk_conf-6-00000002”, “s-local,1”) in new stack
– Goto (sub_nat2int,s-local,1)
– Executing [s-local@sub_nat2int:1] Verbose(“SIP/mdc_trunk_conf-6-00000002”, “1,local”) in new stack
local
– Executing [s-local@sub_nat2int:2] Set(“SIP/mdc_trunk_conf-6-00000002”, “MDC_CALLER_NUM_INTERNAT=004991491718888888”) in new stack
– Executing [s-local@sub_nat2int:3] Return(“SIP/mdc_trunk_conf-6-00000002”, “”) in new stack
– Executing [s@mdc_trunk-3:3] Set(“SIP/mdc_trunk_conf-6-00000002”, “CALLERID(num)=004991491718888888”) in new stack
– Executing [s@mdc_trunk-3:4] Gosub(“SIP/mdc_trunk_conf-6-00000002”, “sub_int2nat,s,1(MDC_CALLER_NUM_NAT,004991491718888888,00,49,0,91)”) in new stack
– Executing [s@sub_int2nat:1] Verbose(“SIP/mdc_trunk_conf-6-00000002”, “1,sub_int2nat:: variable: MDC_CALLER_NUM_NAT - exten: 004991491718888888 - intVAZ: 00 - lkz: 49 - natVAZ: 0 - onKz: 91”) in new stack
sub_int2nat:: variable: MDC_CALLER_NUM_NAT - exten: 004991491718888888 - intVAZ: 00 - lkz: 49 - natVAZ: 0 - onKz: 91
– Executing [s@sub_int2nat:2] GotoIf(“SIP/mdc_trunk_conf-6-00000002”, “0?s-emergency,1”) in new stack
– Executing [s@sub_int2nat:3] GotoIf(“SIP/mdc_trunk_conf-6-00000002”, “0?nat”) in new stack
– Executing [s@sub_int2nat:4] GotoIf(“SIP/mdc_trunk_conf-6-00000002”, “1?s-internat,1”) in new stack
– Goto (sub_int2nat,s-internat,1)
– Executing [s-internat@sub_int2nat:1] Set(“SIP/mdc_trunk_conf-6-00000002”, “MDC_CALLER_NUM_NAT=091491718888888”) in new stack
– Executing [s-internat@sub_int2nat:2] Return(“SIP/mdc_trunk_conf-6-00000002”, “”) in new stack
– Executing [s@mdc_trunk-3:5] Set(“SIP/mdc_trunk_conf-6-00000002”, “CALLERID(num)=091491718888888”) in new stack
– Executing [s@mdc_trunk-3:6] UserEvent(“SIP/mdc_trunk_conf-6-00000002”, “ResolveCallerName,Strategy: default,Outbound: 0,Channel: SIP/mdc_trunk_conf-6-00000002”) in new stack
– Executing [s@mdc_trunk-3:7] Wait(“SIP/mdc_trunk_conf-6-00000002”, “0.25”) in new stack
– Executing [s@mdc_trunk-3:8] Verbose(“SIP/mdc_trunk_conf-6-00000002”, “1,MDC_RESOLVENAME_HITS = 0”) in new stack
MDC_RESOLVENAME_HITS = 0
– Executing [s@mdc_trunk-3:9] Verbose(“SIP/mdc_trunk_conf-6-00000002”, "1,CALLERID(name) = ") in new stack
CALLERID(name) =
– Executing [s@mdc_trunk-3:10] Set(“SIP/mdc_trunk_conf-6-00000002”, “MDC_NUMPREFIX_TRUNK=0”) in new stack
– Executing [s@mdc_trunk-3:11] ExecIf(“SIP/mdc_trunk_conf-6-00000002”, “1?Set(CALLERID(num)=0091491718888888)”) in new stack
– Executing [s@mdc_trunk-3:12] Goto(“SIP/mdc_trunk_conf-6-00000002”, “mdc_mapping-3,4909177777777,1”) in new stack
– Goto (mdc_mapping-3,4909177777777,1)
– Executing [4909177777777@mdc_mapping-3:1] Set(“SIP/mdc_trunk_conf-6-00000002”, “CHANNEL(language)=de”) in new stack
– Executing [4909177777777@mdc_mapping-3:2] Verbose(“SIP/mdc_trunk_conf-6-00000002”, "1,mapping from _%#a-zA-Z0-9]. to 150") in new stack
mapping from _%#a-zA-Z0-9]. to 150
– Executing [4909177777777@mdc_mapping-3:3] Goto(“SIP/mdc_trunk_conf-6-00000002”, “mdc_external,150,1”) in new stack
– Goto (mdc_external,150,1)
– Executing [150@mdc_external:1] SIPAddHeader(“SIP/mdc_trunk_conf-6-00000002”, "“Alert-Info:<http://www.notused.de>;info=alert-external;x-line-id=0"”) in new stack
– Executing [150@mdc_external:2] GosubIf(“SIP/mdc_trunk_conf-6-00000002”, “1?sub_initcall,s,1(ext,150)”) in new stack
– Executing [s@sub_initcall:1] Verbose(“SIP/mdc_trunk_conf-6-00000002”, “1,sub_initcall descent: ext exten: 150”) in new stack
sub_initcall descent: ext exten: 150
– Executing [s@sub_initcall:2] GosubIf(“SIP/mdc_trunk_conf-6-00000002”, “1?sub_initloop,s,1”) in new stack
– Executing [s@sub_initloop:1] Verbose(“SIP/mdc_trunk_conf-6-00000002”, “1,initial loop”) in new stack
initial loop
– Executing [s@sub_initloop:2] Set(“SIP/mdc_trunk_conf-6-00000002”, “MDC_ALIAS_HOP=0”) in new stack
– Executing [s@sub_initloop:3] Return(“SIP/mdc_trunk_conf-6-00000002”, “”) in new stack
– Executing [s@sub_initcall:3] Set(“SIP/mdc_trunk_conf-6-00000002”, “__MDC_TRANSFERBACK_HOP=0”) in new stack
– Executing [s@sub_initcall:4] Set(“SIP/mdc_trunk_conf-6-00000002”, “__MDC_DIALDESCENT=ext”) in new stack
– Executing [s@sub_initcall:5] Goto(“SIP/mdc_trunk_conf-6-00000002”, “ext,1”) in new stack
– Goto (sub_initcall,ext,1)
– Executing [ext@sub_initcall:1] Set(“SIP/mdc_trunk_conf-6-00000002”, “__MDC_DIALCALLERNUMINIT=491718888888”) in new stack
– Executing [ext@sub_initcall:2] Set(“SIP/mdc_trunk_conf-6-00000002”, “__MDC_DIALCALLEENUMINIT=4909177777777”) in new stack
– Executing [ext@sub_initcall:3] Set(“SIP/mdc_trunk_conf-6-00000002”, “__MDC_DIALCALLEENUMMAP=150”) in new stack
– Executing [ext@sub_initcall:4] Return(“SIP/mdc_trunk_conf-6-00000002”, “”) in new stack
– Executing [150@mdc_external:3] Goto(“SIP/mdc_trunk_conf-6-00000002”, “main,150,1”) in new stack
– Goto (main,150,1)
– Executing [150@main:1] Gosub(“SIP/mdc_trunk_conf-6-00000002”, “sub_defcall,s,1(150)”) in new stack
– Executing [s@sub_defcall:1] Set(“SIP/mdc_trunk_conf-6-00000002”, “__MDC_ALIAS_HOP=1”) in new stack
– Executing [s@sub_defcall:2] Set(“SIP/mdc_trunk_conf-6-00000002”, “__MDC_DIALCALLEENUM=150”) in new stack
– Executing [s@sub_defcall:3] Set(“SIP/mdc_trunk_conf-6-00000002”, “__MDC_DIALCALLERNUM=0091491718888888”) in new stack
– Executing [s@sub_defcall:4] GotoIf(“SIP/mdc_trunk_conf-6-00000002”, “1?nozap”) in new stack
– Goto (sub_defcall,s,8)
– Executing [s@sub_defcall:8] Set(“SIP/mdc_trunk_conf-6-00000002”, “__MDC_DIALCALLERNAME=”) in new stack
– Executing [s@sub_defcall:9] Set(“SIP/mdc_trunk_conf-6-00000002”, “__MDC_DIALCHANNELNAME=mdc_trunk_conf-6”) in new stack
– Executing [s@sub_defcall:10] Return(“SIP/mdc_trunk_conf-6-00000002”, “”) in new stack
– Executing [150@main:2] Goto(“SIP/mdc_trunk_conf-6-00000002”, “mdc_distribute,150,1”) in new stack
– Goto (mdc_distribute,150,1)
– Executing [150@mdc_distribute:1] Gosub(“SIP/mdc_trunk_conf-6-00000002”, “sub_user,s,1(callee,1,David.Studnicka,David Studnička,150,150)”) in new stack
– Executing [s@sub_user:1] Verbose(“SIP/mdc_trunk_conf-6-00000002”, “1,sub_user mode callee”) in new stack
sub_user mode callee
– Executing [s@sub_user:2] Goto(“SIP/mdc_trunk_conf-6-00000002”, “s-callee,1”) in new stack
– Goto (sub_user,s-callee,1)
– Executing [s-callee@sub_user:1] Verbose(“SIP/mdc_trunk_conf-6-00000002”, “1,sub_user user id: 1 - user: David.Studnicka - realname: David Studnička - own mailbox: 150 - watched mailbox: 150”) in new stack
sub_user user id: 1 - user: David.Studnicka - realname: David Studnička - own mailbox: 150 - watched mailbox: 150
– Executing [s-callee@sub_user:2] Set(“SIP/mdc_trunk_conf-6-00000002”, “_MDC_CALLEE_USER_ID=1”) in new stack
– Executing [s-callee@sub_user:3] Set(“SIP/mdc_trunk_conf-6-00000002”, “_MDC_CALLEE_ACC_NAME=David.Studnicka”) in new stack
– Executing [s-callee@sub_user:4] Set(“SIP/mdc_trunk_conf-6-00000002”, “_MDC_CALLEE_ACC_REALNAME=David Studnička”) in new stack
– Executing [s-callee@sub_user:5] Set(“SIP/mdc_trunk_conf-6-00000002”, “_MDC_CALLEE_VM_OWN=150”) in new stack
– Executing [s-callee@sub_user:6] Set(“SIP/mdc_trunk_conf-6-00000002”, “_MDC_CALLEE_VM_WATCHED=150”) in new stack
– Executing [s-callee@sub_user:7] Return(“SIP/mdc_trunk_conf-6-00000002”, “”) in new stack
– Executing [150@mdc_distribute:2] Goto(“SIP/mdc_trunk_conf-6-00000002”, “mdc_ident-1,150,1”) in new stack
– Goto (mdc_ident-1,150,1)
– Executing [150@mdc_ident-1:1] NoOp(“SIP/mdc_trunk_conf-6-00000002”, “alias-check:: call forwarding from 150 - 0”) in new stack
– Executing [150@mdc_ident-1:2] GotoIf(“SIP/mdc_trunk_conf-6-00000002”, “0?150-uml,1:150-dial,1”) in new stack
– Goto (mdc_ident-1,150-dial,1)
– Executing [150-dial@mdc_ident-1:1] Set(“SIP/mdc_trunk_conf-6-00000002”, “__MDC_EXTEN=150”) in new stack
– Executing [150-dial@mdc_ident-1:2] Gosub(“SIP/mdc_trunk_conf-6-00000002”, “sub_prefix-150,ext,1”) in new stack
– Executing [ext@sub_prefix-150:1] Verbose(“SIP/mdc_trunk_conf-6-00000002”, “1,no action”) in new stack
no action
– Executing [ext@sub_prefix-150:2] Return(“SIP/mdc_trunk_conf-6-00000002”, “”) in new stack
– Executing [150-dial@mdc_ident-1:3] GotoIf(“SIP/mdc_trunk_conf-6-00000002”, “0?150-unavail,1”) in new stack
– Executing [150-dial@mdc_ident-1:4] GotoIf(“SIP/mdc_trunk_conf-6-00000002”, “0?150-busy,1”) in new stack
– Executing [150-dial@mdc_ident-1:5] Gosub(“SIP/mdc_trunk_conf-6-00000002”, “sub_main-150,ext,1”) in new stack
– Executing [ext@sub_main-150:1] Set(“SIP/mdc_trunk_conf-6-00000002”, “__PICKUPMARK=150”) in new stack
– Executing [ext@sub_main-150:2] Set(“SIP/mdc_trunk_conf-6-00000002”, “__SPYGROUP=150”) in new stack
– Executing [ext@sub_main-150:3] Dial(“SIP/mdc_trunk_conf-6-00000002”, “Local/150@mdc_diallocation”) in new stack
– Called Local/150@mdc_diallocation
– Executing [150@mdc_diallocation:1] Verbose(“Local/150@mdc_diallocation-00000003;2”, “1,dial to 150 for user 1”) in new stack
dial to 150 for user 1
– Executing [150@mdc_diallocation:2] Set(“Local/150@mdc_diallocation-00000003;2”, “__MDC_DIALLOCATION_CHANNEL=Local/150@mdc_diallocation-00000003;2”) in new stack
– Executing [150@mdc_diallocation:3] Set(“Local/150@mdc_diallocation-00000003;2”, “SHARED(MDC_CALLEE_PEERS)=”) in new stack
– Executing [150@mdc_diallocation:4] Set(“Local/150@mdc_diallocation-00000003;2”, “MDC_EXTEN=150”) in new stack
– Executing [150@mdc_diallocation:5] ExecIf(“Local/150@mdc_diallocation-00000003;2”, “0?Set(TMP_TARGET=team):Set(TMP_TARGET=ext)”) in new stack
– Executing [150@mdc_diallocation:6] Gosub(“Local/150@mdc_diallocation-00000003;2”, “sub_split-user,s,1(1)”) in new stack
– Executing [s@sub_split-user:1] Verbose(“Local/150@mdc_diallocation-00000003;2”, “1,get locations for for: 1”) in new stack
get locations for for: 1
– Executing [s@sub_split-user:2] GotoIf(“Local/150@mdc_diallocation-00000003;2”, “0?s-zero,1”) in new stack
– Executing [s@sub_split-user:3] Set(“Local/150@mdc_diallocation-00000003;2”, “TMP_LOCATIONS=1;0”) in new stack
– Executing [s@sub_split-user:4] Set(“Local/150@mdc_diallocation-00000003;2”, “TMP_STATIC_ID=1”) in new stack
– Executing [s@sub_split-user:5] Verbose(“Local/150@mdc_diallocation-00000003;2”, “1,static id: 1”) in new stack
static id: 1
– Executing [s@sub_split-user:6] Set(“Local/150@mdc_diallocation-00000003;2”, “TMP_DYN_ID=0”) in new stack
– Executing [s@sub_split-user:7] Verbose(“Local/150@mdc_diallocation-00000003;2”, “1,dynamic id: 0”) in new stack
dynamic id: 0
– Executing [s@sub_split-user:8] Return(“Local/150@mdc_diallocation-00000003;2”, “”) in new stack
– Executing [150@mdc_diallocation:7] GotoIf(“Local/150@mdc_diallocation-00000003;2”, “0?invalid,1”) in new stack
– Executing [150@mdc_diallocation:8] GotoIf(“Local/150@mdc_diallocation-00000003;2”, “0?dynamic,1”) in new stack
– Executing [150@mdc_diallocation:9] GotoIf(“Local/150@mdc_diallocation-00000003;2”, “1?static,1”) in new stack
– Goto (mdc_diallocation,static,1)
– Executing [static@mdc_diallocation:1] Dial(“Local/150@mdc_diallocation-00000003;2”, “Local/ext@mdc_locallocation-1”) in new stack
– Called Local/ext@mdc_locallocation-1
– Executing [ext@mdc_locallocation-1:1] Set(“Local/ext@mdc_locallocation-1-00000004;2”, “PUSH(SHARED(MDC_CALLEE_PEERS,Local/150@mdc_diallocation-00000003;2))=Local/ext-1@mdc_localdevice-1”) in new stack
– Executing [ext@mdc_locallocation-1:2] Dial(“Local/ext@mdc_locallocation-1-00000004;2”, “Local/ext-1@mdc_localdevice-1”) in new stack
– Called Local/ext-1@mdc_localdevice-1
– Executing [ext-1@mdc_localdevice-1:1] Verbose(“Local/ext-1@mdc_localdevice-1-00000005;2”, “1,Dialing with delay of 0 seconds for 999 seconds”) in new stack

Merkwürdig sehr.

Markus · January 16, 2015, 6:30am

Hi,

Executing [4909177777777@mdc_incoming-6:1] Set(“SIP/mdc_trunk_conf-6-00000002”, “MDC_CALLER_NUM_TRUNK=491718888888”) in new stack

also dein Provider übermittelt als CALLERID(num) - sprich Rufnummer des Anrufers die 491718888888 sprich ohne IntVaz. Im Script, welches die Rufnummern auf ein einheitliches Format bringt, wird die Rufnummer dann als lokale Rufnummer angesehen, sprich eine Rufnummer die sich im selben Ortsnetz befindet. Deshalb werden hier dann die Kennzeichen vorangestellt und die Rufnummer stimmt für den Rückruf nicht mehr. Das könntest du jetzt umgehen indem du beim Amt die IntVAz entfernst, somit wird keine Vereinheitlichung der Rufnummer durchgeführt, folglich sollte der Rückruf dann klappen

– Executing [4909177777777@mdc_incoming-6:2] Set(“SIP/mdc_trunk_conf-6-00000002”, “MDC_CALLEE_NUM_TRUNK=4909177777777”) in new stack

Als Zielrufnummer wird die 4909177777777 übergeben. Die überflüssige 49 hängt wohl mit der Definition beim Amt zusammen, was du bei Durchwahl reg. hinterlegt hast. Überprüfe das bitte mal, ob du einen Wert hier eintragen musst bzw. welchen Wert.

Solltest du den Wert so benötigen, musst du eine zusätzliche eingehende Regeln hinterlegen, welche darauf zutrifft!

Grüße
Markus

Mandarine · January 16, 2015, 9:24am

Habe schon alles ausprobiert, mit Durchwahl.reg, ohne, 09, 49. Es geht aber immer nur, wenn Benutzername und Durchwahl Reg identisch sind.
Die SIPs sind noch aus freenet Zeiten, da habe ich sogar noch max.mustermann Benutzernamen und Durchwahl reg. später war es dann Strato und nun 1und1, mal 09, mal 499 Naja.
Fritzbox, Commander Auerswald, 3cx laufen dort alle out of the box onewallfree, BN, PW, Registrar and thats it. Wir werden also bleiben.

foxpalace · March 22, 2015, 2:03am

Hi - ich möchte mich mal hier einklinken
Ich habe gestern mal ein bissl recherchiert - schlussendlich sieht es ja wohl so aus, dass mit einem vorhandenen 1und1 / Fritzbox der eingehende Anruf nicht möglich ist, da die Fritzbox ins Netz einwählt und den Port 4060 partout nicht freigibt. Die Funktion die Box nur noch als reines DSL-Modem zu nutzen ist mit aktueller Firmeware nicht mehr möglich, da die meisten DSL-Provider ein VLAN für VOIP, IPTV usw. nutzen, welches hier bei 1und1 und Fritzbox automatisch und für den Benutzer unsichtbar genutzt wird.
Die Frage ist also die - welches alternative DSL-Modem gibt (gibts das überhaupt), damit das VLAN auch an der MobyDick nicht “verfällt”, bzw. man den Port 4060 mit all seinen (bisherigen) Vorzügen nutzen kann? Klar, noch ist das Spielerei mit diesem 1und1-Account herum zu experimentieren, aber in Anbetracht von ALL-IP der Telekom, wird das ganze langsam spannend.

Gruß Michael

maik · March 22, 2015, 11:25pm

Hallo Michael,

ich gehe davon aus Du meinst den Port 5060.
Es ist leider so, das die Fritz!Box den Port 5060 leider bei sich terminiert. Einzige Lösung benutze die FritzBox als Mediagateway.

Am 1und1 Anschluss registriert sich die FritzBox, dann richtest Du auf der Box einen SIP User an. Mit diesen Userdaten registrierst Du dann die Mobydick an der FritzBox (als SIP Amt).

Viel Erfolg

Maik

foxpalace · March 22, 2015, 11:44pm

Hallo Maik,

das habe ich auch schon probiert und immer eine einfache Fehlermeldung Anmeldung nicht erfolgreich - Ursache 408 bekommen. Die Fritzbox ist bei mir als Exposed-Host konfiguriert und steht in einem anderen Netz. Dahinter ist ein Linux mit iptables-Firewall - auch den Port zu forwarden hat nichts geholfen.

Gruß Michael

Nachtrag: FritzBox 7320

maik · March 22, 2015, 11:57pm

Hi Michael,

Fehler 408 bedeutet Time Out / Zeitüberschreitung. Eventuell machst Du (testweise) die iptables Firewall für die MobyDick (IP-Adresse) komplett aus.

Grüße

Maik

foxpalace · March 23, 2015, 11:34am

Hi Maik

ich strecke alle viere von mir - ich hab kein Bock mehr
Komplette Firewall aus - forward an - nix
Keine Ahnung - ich geh jetzt ein Bier trinken und lass das VOIP VOIP sein - ist nur komisch, dass ich von intern auf die Fritzbox komme.

Gruß Michael

foxpalace · April 5, 2015, 2:32am

Huhu - ich bin ja so ein Trottel

Da ich doch nur vorkonfigurieren muss und eine Spielwiese brauche, hab ich mir einfach einen 30-tägigen Test-Trunk bei sipgate geholt. Mal schauen, vielleicht bekommt mein Kollege bald Glas, dann wird einfach komplett auf SIP umgestellt.

Gruss Michael

Whissi · April 15, 2015, 8:33am

@ Markus:
Gibt es bzgl. http://community.pascom.net/showthread.php?1278-Probleme-mit-1und1&p=6348&viewfull=1#post6348 schon Fortschritte?

Ich kann selbiges Problem im Zusammenhang mit Telekom VoIP hinter Speedport-Routern beobachten. Irgendwie scheint da bei Inaktivität etwas einzuschlafen (ich vermute auf Seiten des SIP-ALG/siproxd der auf dem Gerät läuft und nicht abschaltbar ist):


mobydick*CLI> sip show registry
Host                                    dnsmgr Username       Refresh State                Reg.Time
tel.t-online.de:5060                    Y      02133XXXXXX        585 Registered           Wed, 15 Apr 2015 16:50:45
1 SIP registrations.

[Apr 15 16:52:15] NOTICE[2288]: chan_sip.c:23346 handle_response_register: Outbound Registration: Expiry for tel.t-online.de is 600 sec (Scheduling reregistration in 585 s)
Really destroying SIP dialog '01fba824420a95e01fa978fb3f12b14c@127.0.0.1' Method: REGISTER
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
    -- Executing [02133XXXXXX@no-auth-in:1] Gosub("SIP/tel.t-online.de-00000012", "sub_emergency-check,s,1(02133XXXXXX)") in new stack
    -- Executing [s@sub_emergency-check:1] Verbose("SIP/tel.t-online.de-00000012", "1,sub_emergency-check:: exten: 02133XXXXXX - descent: ") in new stack
 sub_emergency-check:: exten: 02133XXXXXX - descent:
    -- Executing [s@sub_emergency-check:2] GotoIf("SIP/tel.t-online.de-00000012", "1?02133XXXXXX,1") in new stack
    -- Goto (sub_emergency-check,02133XXXXXX,1)
    -- Channel 'SIP/tel.t-online.de-00000012' sent to invalid extension: context,exten,priority=sub_emergency-check,02133XXXXXX,1
    -- Executing * Return("SIP/tel.t-online.de-00000012", "") in new stack
    -- Executing [02133XXXXXX@no-auth-in:2] GotoIf("SIP/tel.t-online.de-00000012", "0?mdc_emergency,dial,1:mdc_emergency,invalid,1") in new stack
    -- Goto (mdc_emergency,invalid,1)
    -- Executing [invalid@mdc_emergency:1] NoOp("SIP/tel.t-online.de-00000012", "mdc_emergency::  is no emergency call") in new stack
    -- Executing [invalid@mdc_emergency:2] Answer("SIP/tel.t-online.de-00000012", "") in new stack
    -- Executing [invalid@mdc_emergency:3] Playback("SIP/tel.t-online.de-00000012", "beeperr") in new stack
    -- <SIP/tel.t-online.de-00000012> Playing 'beeperr.slin' (language 'en')
    -- Executing [invalid@mdc_emergency:4] Hangup("SIP/tel.t-online.de-00000012", "20") in new stack
  == Spawn extension (mdc_emergency, invalid, 4) exited non-zero on 'SIP/tel.t-online.de-00000012'
[Apr 15 16:55:19] NOTICE[2288][C-0000000c]: chan_sip.c:25712 handle_request_invite: Unable to create/find SIP channel for this INVITE
[Apr 15 16:55:49] WARNING[2288]: chan_sip.c:4037 retrans_pkt: Retransmission timeout reached on transmission p65542t1429109838m116690c60828270s2 for seqno 1 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32000ms with no response

2min später mag dann wieder alles funktionieren, ohne manuellen Eingriff.


Anrufer: +4930YYYYYYYY
   Ziel: +492133XXXXXXXX

              IP MD: 192.168.2.99
IP Speedport Router: 192.168.2.1

         Externe IP: 87.170.81.138

Normalerweise sehen eingehende Telekom-Anrufe in sipgrep hier wie folgt aus:


U 2015/04/15 16:04:16.522112 217.0.23.8:5060 -> 192.168.2.99:5060
INVITE sip:02133XXXXXX@87.170.81.138:5060 SIP/2.0.
Via: SIP/2.0/UDP 217.0.23.8:5060;branch=z9hG4bKg3Zqkv7ixcfs4ipyyc3ilnqomv6b5gmyq.
Record-Route: <sip:217.0.23.8;transport=udp;lr>.
From: <sip:+4930YYYYYYYY@tel.t-online.de;user=phone>;tag=h7g4Esbg_p65542t1429106656m709819c60572232s1_3533613112-1731897868.
To: <sip:02133XXXXXX@tel.t-online.de>.
Call-ID: p65542t1429106656m709819c60572232s2.
CSeq: 1 INVITE.
Contact: <sip:sgc_c@217.0.23.8;transport=udp>;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel".
max-forwards: 62.
accept-contact: *;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel".
min-se: 900.
p-asserted-identity: <sip:+4930YYYYYYYY@tel.t-online.de;user=phone>.
session-expires: 1800.
supported: timer.
supported: 100rel.
supported: histinfo.
session-id: 107312f31273cc3cfb40ce279e2a952d.
Allow: REGISTER, REFER, NOTIFY, SUBSCRIBE, INFO, PRACK, PUBLISH, UPDATE, INVITE, ACK, OPTIONS, CANCEL, BYE.
Content-Type: application/sdp.
Content-Length:   357.
.
v=0.
o=- 1983188913 3533612605 IN IP4 192.168.2.1.
s=call.
c=IN IP4 192.168.2.1.
t=0 0.
m=audio 7070 RTP/AVP 8 0 2 102 100 99 97 101.
a=sendrecv.
a=rtpmap:2 G726-32/8000.
a=rtpmap:102 G726-32/8000.
a=rtpmap:100 G726-40/8000.
a=rtpmap:99 G726-24/8000.
a=rtpmap:97 iLBC/8000.
a=fmtp:97 mode=30.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=ptime:20.


U 2015/04/15 16:04:16.524256 192.168.2.99:5060 -> 217.0.23.8:5060
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP 217.0.23.8:5060;branch=z9hG4bKg3Zqkv7ixcfs4ipyyc3ilnqomv6b5gmyq;received=217.0.23.8.
Record-Route: <sip:217.0.23.8;transport=udp;lr>.
From: <sip:+4930YYYYYYYY@tel.t-online.de;user=phone>;tag=h7g4Esbg_p65542t1429106656m709819c60572232s1_3533613112-1731897868.
To: <sip:02133XXXXXX@tel.t-online.de>.
Call-ID: p65542t1429106656m709819c60572232s2.
CSeq: 1 INVITE.
Server: Asterisk PBX 11.6-cert10.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
Supported: replaces, timer.
Session-Expires: 1800;refresher=uas.
Contact: <sip:02133XXXXXX@87.170.81.138:5060>.
Content-Length: 0.
.

Anrufe werden also von “außen” signalisiert. Der Speedport hat sich lediglich die Audiostream gegriffen.

Wenn es eben mal nicht funktioniert, dann erfolgt die Signalisierung auch über den Speedport:


U 2015/04/15 15:54:33.202192 192.168.2.1:56005 -> 192.168.2.99:5060
INVITE sip:02133XXXXXXXX@87.170.81.138:5060 SIP/2.0.
Via: SIP/2.0/UDP 217.0.23.8:5060;branch=z9hG4bKg3Zqkv7imio3s4iiwshfw7vpkgi3s9uk6.
Record-Route: <sip:217.0.23.8;transport=udp;lr>.
From: <sip:+4930YYYYYYYY@tel.t-online.de;user=phone>;tag=h7g4Esbg_p65542t1429106073m379220c60532507s1_2950283138-1450134101.
To: <sip:02133XXXXXXXX@tel.t-online.de>.
Call-ID: p65542t1429106073m379220c60532507s2.
CSeq: 1 INVITE.
Contact: <sip:sgc_c@217.0.23.8;transport=udp>;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel".
max-forwards: 62.
accept-contact: *;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel".
min-se: 900.
p-asserted-identity: <sip:+4930YYYYYYYY@tel.t-online.de;user=phone>.
session-expires: 1800.
supported: timer.
supported: 100rel.
supported: histinfo.
session-id: df30cf34ff38c43cfa40ce27a02a972d.
Allow: REGISTER, REFER, NOTIFY, SUBSCRIBE, INFO, PRACK, PUBLISH, UPDATE, INVITE, ACK, OPTIONS, CANCEL, BYE.
Content-Type: application/sdp.
Content-Length:   357.
.
v=0.
o=- 1757369526 2950282579 IN IP4 192.168.2.1.
s=call.
c=IN IP4 192.168.2.1.
t=0 0.
m=audio 7070 RTP/AVP 8 0 2 102 100 99 97 101.
a=sendrecv.
a=rtpmap:2 G726-32/8000.
a=rtpmap:102 G726-32/8000.
a=rtpmap:100 G726-40/8000.
a=rtpmap:99 G726-24/8000.
a=rtpmap:97 iLBC/8000.
a=fmtp:97 mode=30.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=ptime:20.


U 2015/04/15 15:54:33.203895 192.168.2.99:5060 -> 192.168.2.1:5060
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP 217.0.23.8:5060;branch=z9hG4bKg3Zqkv7imio3s4iiwshfw7vpkgi3s9uk6;received=192.168.2.1.
Record-Route: <sip:217.0.23.8;transport=udp;lr>.
From: <sip:+4930YYYYYYYY@tel.t-online.de;user=phone>;tag=h7g4Esbg_p65542t1429106073m379220c60532507s1_2950283138-1450134101.
To: <sip:02133XXXXXXXX@tel.t-online.de>.
Call-ID: p65542t1429106073m379220c60532507s2.
CSeq: 1 INVITE.
Server: Asterisk PBX 11.6-cert10.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
Supported: replaces, timer.
Session-Expires: 1800;refresher=uas.
Contact: <sip:02133XXXXXXXX@192.168.2.99:5060>.
Content-Length: 0.
.


U 2015/04/15 15:54:33.205603 192.168.2.99:5060 -> 192.168.2.1:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 217.0.23.8:5060;branch=z9hG4bKg3Zqkv7imio3s4iiwshfw7vpkgi3s9uk6;received=192.168.2.1.
Record-Route: <sip:217.0.23.8;transport=udp;lr>.
From: <sip:+4930YYYYYYYY@tel.t-online.de;user=phone>;tag=h7g4Esbg_p65542t1429106073m379220c60532507s1_2950283138-1450134101.
To: <sip:02133XXXXXXXX@tel.t-online.de>;tag=as02e8f3f5.
Call-ID: p65542t1429106073m379220c60532507s2.
CSeq: 1 INVITE.
Server: Asterisk PBX 11.6-cert10.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
Supported: replaces, timer.
Session-Expires: 1800;refresher=uas.
Contact: <sip:02133XXXXXXXX@192.168.2.99:5060>.
Content-Type: application/sdp.
Require: timer.
Content-Length: 264.
.
v=0.
o=root 2039367919 2039367919 IN IP4 192.168.2.99.
s=Asterisk PBX 11.6-cert10.
c=IN IP4 192.168.2.99.
t=0 0.
m=audio 40088 RTP/AVP 0 8 101.
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 2015/04/15 15:54:33.704218 192.168.2.1:56005 -> 192.168.2.99:5060
INVITE sip:02133XXXXXXXX@87.170.81.138:5060 SIP/2.0.
Via: SIP/2.0/UDP 217.0.23.8:5060;branch=z9hG4bKg3Zqkv7imio3s4iiwshfw7vpkgi3s9uk6.
Record-Route: <sip:217.0.23.8;transport=udp;lr>.
From: <sip:+4930YYYYYYYY@tel.t-online.de;user=phone>;tag=h7g4Esbg_p65542t1429106073m379220c60532507s1_2950283138-1450134101.
To: <sip:02133XXXXXXXX@tel.t-online.de>.
Call-ID: p65542t1429106073m379220c60532507s2.
CSeq: 1 INVITE.
Contact: <sip:sgc_c@217.0.23.8;transport=udp>;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel".
max-forwards: 62.
accept-contact: *;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel".
min-se: 900.
p-asserted-identity: <sip:+4930YYYYYYYY@tel.t-online.de;user=phone>.
session-expires: 1800.
supported: timer.
supported: 100rel.
supported: histinfo.
session-id: df30cf34ff38c43cfa40ce27a02a972d.
Allow: REGISTER, REFER, NOTIFY, SUBSCRIBE, INFO, PRACK, PUBLISH, UPDATE, INVITE, ACK, OPTIONS, CANCEL, BYE.
Content-Type: application/sdp.
Content-Length:   357.
.
v=0.
o=- 1757369526 2950282579 IN IP4 192.168.2.1.
s=call.
c=IN IP4 192.168.2.1.
t=0 0.
m=audio 7070 RTP/AVP 8 0 2 102 100 99 97 101.
a=sendrecv.
a=rtpmap:2 G726-32/8000.
a=rtpmap:102 G726-32/8000.
a=rtpmap:100 G726-40/8000.
a=rtpmap:99 G726-24/8000.
a=rtpmap:97 iLBC/8000.
a=fmtp:97 mode=30.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=ptime:20.


U 2015/04/15 15:54:33.704444 192.168.2.99:5060 -> 192.168.2.1:5060
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP 217.0.23.8:5060;branch=z9hG4bKg3Zqkv7imio3s4iiwshfw7vpkgi3s9uk6;received=192.168.2.1.
Record-Route: <sip:217.0.23.8;transport=udp;lr>.
From: <sip:+4930YYYYYYYY@tel.t-online.de;user=phone>;tag=h7g4Esbg_p65542t1429106073m379220c60532507s1_2950283138-1450134101.
To: <sip:02133XXXXXXXX@tel.t-online.de>.
Call-ID: p65542t1429106073m379220c60532507s2.
CSeq: 1 INVITE.
Server: Asterisk PBX 11.6-cert10.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
Supported: replaces, timer.
Session-Expires: 1800;refresher=uas.
Contact: <sip:02133XXXXXXXX@192.168.2.99:5060>.
Content-Length: 0.
.


U 2015/04/15 15:54:33.704662 192.168.2.99:5060 -> 192.168.2.1:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 217.0.23.8:5060;branch=z9hG4bKg3Zqkv7imio3s4iiwshfw7vpkgi3s9uk6;received=192.168.2.1.
Record-Route: <sip:217.0.23.8;transport=udp;lr>.
From: <sip:+492133YYYYYYYY@tel.t-online.de;user=phone>;tag=h7g4Esbg_p65542t1429106073m379220c60532507s1_2950283138-1450134101.
To: <sip:02133XXXXXXXX@tel.t-online.de>;tag=as02e8f3f5.
Call-ID: p65542t1429106073m379220c60532507s2.
CSeq: 1 INVITE.
Server: Asterisk PBX 11.6-cert10.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
Supported: replaces, timer.
Session-Expires: 1800;refresher=uas.
Contact: <sip:02133XXXXXXXX@192.168.2.99:5060>.
Content-Type: application/sdp.
Require: timer.
Content-Length: 264.
.
v=0.
o=root 2039367919 2039367920 IN IP4 192.168.2.99.
s=Asterisk PBX 11.6-cert10.
c=IN IP4 192.168.2.99.
t=0 0.
m=audio 40088 RTP/AVP 0 8 101.
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 2015/04/15 15:54:33.704781 192.168.2.99:5060 -> 192.168.2.1:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 217.0.23.8:5060;branch=z9hG4bKg3Zqkv7imio3s4iiwshfw7vpkgi3s9uk6;received=192.168.2.1.
Record-Route: <sip:217.0.23.8;transport=udp;lr>.
From: <sip:+4930YYYYYYYY@tel.t-online.de;user=phone>;tag=h7g4Esbg_p65542t1429106073m379220c60532507s1_2950283138-1450134101.
To: <sip:02133XXXXXXXX@tel.t-online.de>;tag=as02e8f3f5.
Call-ID: p65542t1429106073m379220c60532507s2.
CSeq: 1 INVITE.
Server: Asterisk PBX 11.6-cert10.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
Supported: replaces, timer.
Session-Expires: 1800;refresher=uas.
Contact: <sip:02133XXXXXXXX@192.168.2.99:5060>.
Content-Type: application/sdp.
Require: timer.
Content-Length: 264.
.
v=0.
o=root 2039367919 2039367919 IN IP4 192.168.2.99.
s=Asterisk PBX 11.6-cert10.
c=IN IP4 192.168.2.99.
t=0 0.
m=audio 40088 RTP/AVP 0 8 101.
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.

...das wiederholt sich nun, was ich aktuell nicht verstehe, durch das no-auth-in sollte die Sache doch eigentlich terminiert werden?!

…aber das scheitert dann eben mit oben genannten Fehler.*