Sipgate Trunk Peer unreachable

ulf.kosack · April 26, 2014, 8:46am

Hallo zusammen,

ich versuche gerade mit einer Community-Version (v7.06.03) eine Testumgebung aufzubauen. Dazu habe ich auch einen Sipgate Trunk-Anschluss konfiguriert. Leider bekomme ich nur eingehende Verbindungen hin. Bei ausgehenden Verbindungen erhalte ich immer ein “Temporarily Unavailable”. Im Asterisk-Debug erscheint dazu folgendes:

Connected to Asterisk 11.6-cert1 currently running on mobydick (pid = 3899)
  == Using SIP RTP CoS mark 5
    -- Executing [02286896297@mdc_location-3:1] Verbose("SIP/HJFEW0FDm712e96-00000002", "1,Ursula Borrmann") in new stack
 Ursula Borrmann
    -- Executing [02286896297@mdc_location-3:2] Set("SIP/HJFEW0FDm712e96-00000002", "__MDC_CALLER_LOCATION_ID=3") in new stack
    -- Executing [02286896297@mdc_location-3:3] Set("SIP/HJFEW0FDm712e96-00000002", "__MDC_CALLER_LOCATION_NAME=Ursula Borrmann") in new stack
    -- Executing [02286896297@mdc_location-3:4] Gosub("SIP/HJFEW0FDm712e96-00000002", "sub_location,s,1(3,02286896297)") in new stack
    -- Executing [s@sub_location:1] Verbose("SIP/HJFEW0FDm712e96-00000002", "1,used location id: 3 - dialed extension: 02286896297") in new stack
 used location id: 3 - dialed extension: 02286896297
    -- Executing [s@sub_location:2] Set("SIP/HJFEW0FDm712e96-00000002", "MDC_CALLEE_NUM=02286896297") in new stack
    -- Executing [s@sub_location:3] Return("SIP/HJFEW0FDm712e96-00000002", "") in new stack
    -- Executing [02286896297@mdc_location-3:5] Gosub("SIP/HJFEW0FDm712e96-00000002", "sub_split-location,s,1(3)") in new stack
    -- Executing [s@sub_split-location:1] Verbose("SIP/HJFEW0FDm712e96-00000002", "1,get user for location: 3") in new stack
 get user for location: 3
    -- Executing [s@sub_split-location:2] GotoIf("SIP/HJFEW0FDm712e96-00000002", "0?s-zero,1") in new stack
    -- Executing [s@sub_split-location:3] Set("SIP/HJFEW0FDm712e96-00000002", "TMP_USER_ID=4") in new stack
    -- Executing [s@sub_split-location:4] Verbose("SIP/HJFEW0FDm712e96-00000002", "1,user id: 4") in new stack
 user id: 4
    -- Executing [s@sub_split-location:5] Return("SIP/HJFEW0FDm712e96-00000002", "") in new stack
    -- Executing [02286896297@mdc_location-3:6] Goto("SIP/HJFEW0FDm712e96-00000002", "mdc_user-4,02286896297,1") in new stack
    -- Goto (mdc_user-4,02286896297,1)
    -- Executing [02286896297@mdc_user-4:1] Verbose("SIP/HJFEW0FDm712e96-00000002", "1,uborrmann") in new stack
 uborrmann
    -- Executing [02286896297@mdc_user-4:2] Gosub("SIP/HJFEW0FDm712e96-00000002", "sub_user,s,1(caller,4,32,uborrmann,Ursula Borrmann,32,32,32)") in new stack
    -- Executing [s@sub_user:1] Verbose("SIP/HJFEW0FDm712e96-00000002", "1,sub_user mode caller") in new stack
 sub_user mode caller
    -- Executing [s@sub_user:2] Goto("SIP/HJFEW0FDm712e96-00000002", "s-caller,1") in new stack
    -- Goto (sub_user,s-caller,1)
    -- Executing [s-caller@sub_user:1] Verbose("SIP/HJFEW0FDm712e96-00000002", "1,sub_user user id: 4 - user exten: 32 - user: uborrmann - realname: Ursula Borrmann - own mailbox: 32 - watched mailbox: 32 - callerid(num): 32") in new stack
 sub_user user id: 4 - user exten: 32 - user: uborrmann - realname: Ursula Borrmann - own mailbox: 32 - watched mailbox: 32 - callerid(num): 32
    -- Executing [s-caller@sub_user:2] Set("SIP/HJFEW0FDm712e96-00000002", "__MDC_CALLER_USER_ID=4") in new stack
    -- Executing [s-caller@sub_user:3] Set("SIP/HJFEW0FDm712e96-00000002", "__MDC_CALLER_NUM=32") in new stack
    -- Executing [s-caller@sub_user:4] Set("SIP/HJFEW0FDm712e96-00000002", "__MDC_CALLER_ACC_NAME=uborrmann") in new stack
    -- Executing [s-caller@sub_user:5] Set("SIP/HJFEW0FDm712e96-00000002", "__MDC_CALLER_ACC_REALNAME=Ursula Borrmann") in new stack
    -- Executing [s-caller@sub_user:6] Set("SIP/HJFEW0FDm712e96-00000002", "__MDC_CALLER_VM_OWN=32") in new stack
    -- Executing [s-caller@sub_user:7] Set("SIP/HJFEW0FDm712e96-00000002", "__MDC_CALLER_VM_WATCHED=32") in new stack
    -- Executing [s-caller@sub_user:8] Set("SIP/HJFEW0FDm712e96-00000002", "CALLERID(name)=Ursula Borrmann") in new stack
    -- Executing [s-caller@sub_user:9] Set("SIP/HJFEW0FDm712e96-00000002", "CALLERID(num)=32") in new stack
    -- Executing [s-caller@sub_user:10] Return("SIP/HJFEW0FDm712e96-00000002", "") in new stack
    -- Executing [02286896297@mdc_user-4:3] Goto("SIP/HJFEW0FDm712e96-00000002", "internal,02286896297,1") in new stack
    -- Goto (internal,02286896297,1)
    -- Executing [02286896297@internal:1] SIPAddHeader("SIP/HJFEW0FDm712e96-00000002", ""Alert-Info:<http://www.notused.de>;info=alert-internal;x-line-id=0"") in new stack
    -- Executing [02286896297@internal:2] GosubIf("SIP/HJFEW0FDm712e96-00000002", "1?sub_initcall,s,1(int,02286896297)") in new stack
    -- Executing [s@sub_initcall:1] Verbose("SIP/HJFEW0FDm712e96-00000002", "1,sub_initcall descent: int exten: 02286896297") in new stack
 sub_initcall descent: int exten: 02286896297
    -- Executing [s@sub_initcall:2] GosubIf("SIP/HJFEW0FDm712e96-00000002", "1?sub_initloop,s,1") in new stack
    -- Executing [s@sub_initloop:1] Verbose("SIP/HJFEW0FDm712e96-00000002", "1,initial loop") in new stack
 initial loop
    -- Executing [s@sub_initloop:2] Set("SIP/HJFEW0FDm712e96-00000002", "MDC_ALIAS_HOP=0") in new stack
    -- Executing [s@sub_initloop:3] Return("SIP/HJFEW0FDm712e96-00000002", "") in new stack
    -- Executing [s@sub_initcall:3] Set("SIP/HJFEW0FDm712e96-00000002", "__MDC_DIALDESCENT=int") in new stack
    -- Executing [s@sub_initcall:4] Goto("SIP/HJFEW0FDm712e96-00000002", "int,1") in new stack
    -- Goto (sub_initcall,int,1)
    -- Executing [int@sub_initcall:1] Set("SIP/HJFEW0FDm712e96-00000002", "__MDC_DIALCALLERNUMINIT=32") in new stack
    -- Executing [int@sub_initcall:2] Set("SIP/HJFEW0FDm712e96-00000002", "__MDC_DIALCALLEENUMINIT=02286896297") in new stack
    -- Executing [int@sub_initcall:3] Return("SIP/HJFEW0FDm712e96-00000002", "") in new stack
    -- Executing [02286896297@internal:3] Goto("SIP/HJFEW0FDm712e96-00000002", "main,02286896297,1") in new stack
    -- Goto (main,02286896297,1)
    -- Executing [02286896297@main:1] Gosub("SIP/HJFEW0FDm712e96-00000002", "sub_defcall,s,1(02286896297)") in new stack
    -- Executing [s@sub_defcall:1] Set("SIP/HJFEW0FDm712e96-00000002", "__MDC_ALIAS_HOP=1") in new stack
    -- Executing [s@sub_defcall:2] Set("SIP/HJFEW0FDm712e96-00000002", "__MDC_DIALCALLEENUM=02286896297") in new stack
    -- Executing [s@sub_defcall:3] Set("SIP/HJFEW0FDm712e96-00000002", "__MDC_DIALCALLERNUM=32") in new stack
    -- Executing [s@sub_defcall:4] Set("SIP/HJFEW0FDm712e96-00000002", "__MDC_DIALCHANNELNAME=HJFEW0FDm712e96") in new stack
    -- Executing [s@sub_defcall:5] GotoIf("SIP/HJFEW0FDm712e96-00000002", "1?nozap") in new stack
    -- Goto (sub_defcall,s,8)
    -- Executing [s@sub_defcall:8] Set("SIP/HJFEW0FDm712e96-00000002", "__MDC_DIALCALLERNAME=Ursula Borrmann") in new stack
    -- Executing [s@sub_defcall:9] Return("SIP/HJFEW0FDm712e96-00000002", "") in new stack
    -- Executing [02286896297@main:2] Goto("SIP/HJFEW0FDm712e96-00000002", "mdc_distribute,02286896297,1") in new stack
    -- Goto (mdc_distribute,02286896297,1)
    -- Executing [02286896297@mdc_distribute:1] Set("SIP/HJFEW0FDm712e96-00000002", "MDC_INPREFIX_TRUNK=") in new stack
    -- Executing [02286896297@mdc_distribute:2] Verbose("SIP/HJFEW0FDm712e96-00000002", "1,remove inprefix: ") in new stack
 remove inprefix:
    -- Executing [02286896297@mdc_distribute:3] Goto("SIP/HJFEW0FDm712e96-00000002", "mdc_outgoing-3,02286896297,1") in new stack
    -- Goto (mdc_outgoing-3,02286896297,1)
    -- Executing [02286896297@mdc_outgoing-3:1] Gosub("SIP/HJFEW0FDm712e96-00000002", "sub_trunk-outgoing-3,02286896297,1") in new stack
    -- Executing [02286896297@sub_trunk-outgoing-3:1] Gosub("SIP/HJFEW0FDm712e96-00000002", "sub_nat2int,s,1(MDC_CALLEE_NUM_INTERNAT,02286896297,00,49,0,228)") in new stack
    -- Executing [s@sub_nat2int:1] Verbose("SIP/HJFEW0FDm712e96-00000002", "1,sub_nat2int:: variable: MDC_CALLEE_NUM_INTERNAT - CALLERID(num): 02286896297 - intVAZ: 00 - lkz: 49 - natVAZ: 0 - onKz: 228") in new stack
 sub_nat2int:: variable: MDC_CALLEE_NUM_INTERNAT - CALLERID(num): 02286896297 - intVAZ: 00 - lkz: 49 - natVAZ: 0 - onKz: 228
    -- Executing [s@sub_nat2int:2] GotoIf("SIP/HJFEW0FDm712e96-00000002", "0?s-int,1") in new stack
    -- Executing [s@sub_nat2int:3] GotoIf("SIP/HJFEW0FDm712e96-00000002", "0?s-int,1") in new stack
    -- Executing [s@sub_nat2int:4] GotoIf("SIP/HJFEW0FDm712e96-00000002", "0?s-intshort,1") in new stack
    -- Executing [s@sub_nat2int:5] GotoIf("SIP/HJFEW0FDm712e96-00000002", "0?s-natshort,1") in new stack
    -- Executing [s@sub_nat2int:6] GotoIf("SIP/HJFEW0FDm712e96-00000002", "1?s-nat,1") in new stack
    -- Goto (sub_nat2int,s-nat,1)
    -- Executing [s-nat@sub_nat2int:1] Verbose("SIP/HJFEW0FDm712e96-00000002", "1,national") in new stack
 national
    -- Executing [s-nat@sub_nat2int:2] Set("SIP/HJFEW0FDm712e96-00000002", "MDC_CALLEE_NUM_INTERNAT=00492286896297") in new stack
    -- Executing [s-nat@sub_nat2int:3] Return("SIP/HJFEW0FDm712e96-00000002", "") in new stack
    -- Executing [02286896297@sub_trunk-outgoing-3:2] Gosub("SIP/HJFEW0FDm712e96-00000002", "sub_int2nat,s,1(MDC_CALLEE_NUM_NAT,00492286896297,00,49,0,228)") in new stack
    -- Executing [s@sub_int2nat:1] Verbose("SIP/HJFEW0FDm712e96-00000002", "1,sub_int2nat:: variable: MDC_CALLEE_NUM_NAT - exten: 00492286896297 - intVAZ: 00 - lkz: 49 - natVAZ: 0 - onKz: 228") in new stack
 sub_int2nat:: variable: MDC_CALLEE_NUM_NAT - exten: 00492286896297 - intVAZ: 00 - lkz: 49 - natVAZ: 0 - onKz: 228
    -- Executing [s@sub_int2nat:2] GotoIf("SIP/HJFEW0FDm712e96-00000002", "0?nat") in new stack
    -- Executing [s@sub_int2nat:3] GotoIf("SIP/HJFEW0FDm712e96-00000002", "1?s-internat,1") in new stack
    -- Goto (sub_int2nat,s-internat,1)
    -- Executing [s-internat@sub_int2nat:1] Set("SIP/HJFEW0FDm712e96-00000002", "MDC_CALLEE_NUM_NAT=02286896297") in new stack
    -- Executing [s-internat@sub_int2nat:2] Return("SIP/HJFEW0FDm712e96-00000002", "") in new stack
    -- Executing [02286896297@sub_trunk-outgoing-3:3] SIPAddHeader("SIP/HJFEW0FDm712e96-00000002", "P-Preferred-Identity: sip:4922828695132@sipconnect.sipgate.de") in new stack
    -- Executing [02286896297@sub_trunk-outgoing-3:4] UserEvent("SIP/HJFEW0FDm712e96-00000002", "ResolveCallerName,Strategy: default,Outbound: 1,Channel: SIP/HJFEW0FDm712e96-00000002") in new stack
    -- Executing [02286896297@sub_trunk-outgoing-3:5] Wait("SIP/HJFEW0FDm712e96-00000002", "0.25") in new stack
    -- Executing [02286896297@sub_trunk-outgoing-3:6] Verbose("SIP/HJFEW0FDm712e96-00000002", "1,MDC_RESOLVENAME_HITS = 0") in new stack
 MDC_RESOLVENAME_HITS = 0
    -- Executing [02286896297@sub_trunk-outgoing-3:7] Verbose("SIP/HJFEW0FDm712e96-00000002", "1,MDC_DIALCALLEENAME = ") in new stack
 MDC_DIALCALLEENAME =
    -- Executing [02286896297@sub_trunk-outgoing-3:8] Verbose("SIP/HJFEW0FDm712e96-00000002", "1,sipgate - Standardtelefonie") in new stack
 sipgate - Standardtelefonie
    -- Executing [02286896297@sub_trunk-outgoing-3:9] Dial("SIP/HJFEW0FDm712e96-00000002", "SIP/mdc_trunk_conf-3/02286896297,,t") in new stack
[Apr 26 17:25:37] WARNING[4482][C-00000002]: 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 [02286896297@sub_trunk-outgoing-3:10] Return("SIP/HJFEW0FDm712e96-00000002", "") in new stack
    -- Executing [02286896297@mdc_outgoing-3:2] Hangup("SIP/HJFEW0FDm712e96-00000002", "19") in new stack
  == Spawn extension (mdc_outgoing-3, 02286896297, 2) exited non-zero on 'SIP/HJFEW0FDm712e96-00000002'
    -- Executing [h@mdc_outgoing-3:1] Gosub("SIP/HJFEW0FDm712e96-00000002", "def_hangup,s,1(,CHANUNAVAIL,,CALL)") in new stack
    -- Executing [s@def_hangup:1] NoOp("SIP/HJFEW0FDm712e96-00000002", ">>>def_hangup:: EXTEN:  DIALSTATUS: CHANUNAVAIL QUEUESTATUS:  REASON: CALL") in new stack

Ich habe auch schon die Threads Sip peer und Voipgate peer und Verbindungsproblem mit Sipgate gelesen. Dementsprechend habe ich die folgenden Ausgaben überprüft:

mobydick*CLI> sip show registry
Host                                    dnsmgr Username       Refresh State                Reg.Time
sipconnect.sipgate.de:5060              N      XXXXXXXt0          585 Registered           Sat, 26 Apr 2014 17:26:07
1 SIP registrations.

mobydick*CLI> sip show peers
Name/username             Host                                    Dyn Forcerport ACL Port     Status      Description
HJFEW0FDm712e96/HJFEW0FDm 192.168.200.31                           D   a             5060     OK (5 ms)
InSnMc2gV3fb78a/InSnMc2gV 192.168.200.41                           D   a             5060     OK (21 ms)
LGPhone                   (Unspecified)                            D   a             0        UNKNOWN
mdc_trunk_conf-3/XXXXXXXt 217.10.68.150                                a             5060     UNREACHABLE
4 sip peers [Monitored: 2 online, 2 offline Unmonitored: 0 online, 0 offline]

Der State=Registered scheint wiederzuspiegeln, dass auf der Sipgate Management-Seite der Anschluss als online geführt wird. Nur den Peer erreichen kann ich nicht.

Dementsprechend habe ich die Konfiguration unter sys/asterisk/configure/sip/file um die Parameter externip, localnet, nat und transport erweitert, wie es in den beiden o.g. Threads beschrieben wurde.

Leider bringt das auch keine Besserung

.

Die Anlage läuft virtuell auf VMWare hinter einem LANCOM Router, der den Internet-Anschluss bereitstellt. Auf der Firewall konnte ich keine Blockierungen erkennen.

Mit “sip set debug peer mdc_trunk_conf-3” bekomme ich folgende Ausgabe:

<--- SIP read from UDP:217.10.68.150:5060 --->

<------------->
Reliably Transmitting (no NAT) to 217.10.68.150:5060:
OPTIONS sip:sipconnect.sipgate.de SIP/2.0
Via: SIP/2.0/UDP 192.168.1.251:5060;branch=z9hG4bK3fad19ce
Max-Forwards: 70
From: "asterisk" <sip:XXXXXXXt0@192.168.1.251>;tag=as314b9aa4
To: <sip:sipconnect.sipgate.de>
Contact: <sip:XXXXXXXt0@192.168.1.251:5060>
Call-ID: 070ee473671582cc67a8cc7c5622ddd3@192.168.1.251:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.6-cert1
Date: Sat, 26 Apr 2014 15:38:32 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
Retransmitting #1 (no NAT) to 217.10.68.150:5060:
OPTIONS sip:sipconnect.sipgate.de SIP/2.0
Via: SIP/2.0/UDP 192.168.1.251:5060;branch=z9hG4bK3fad19ce
Max-Forwards: 70
From: "asterisk" <sip:XXXXXXXt0@192.168.1.251>;tag=as314b9aa4
To: <sip:sipconnect.sipgate.de>
Contact: <sip:XXXXXXXt0@192.168.1.251:5060>
Call-ID: 070ee473671582cc67a8cc7c5622ddd3@192.168.1.251:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.6-cert1
Date: Sat, 26 Apr 2014 15:38:32 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
Retransmitting #2 (no NAT) to 217.10.68.150:5060:
OPTIONS sip:sipconnect.sipgate.de SIP/2.0
Via: SIP/2.0/UDP 192.168.1.251:5060;branch=z9hG4bK3fad19ce
Max-Forwards: 70
From: "asterisk" <sip:XXXXXXXt0@192.168.1.251>;tag=as314b9aa4
To: <sip:sipconnect.sipgate.de>
Contact: <sip:XXXXXXXt0@192.168.1.251:5060>
Call-ID: 070ee473671582cc67a8cc7c5622ddd3@192.168.1.251:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.6-cert1
Date: Sat, 26 Apr 2014 15:38:32 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
Retransmitting #3 (no NAT) to 217.10.68.150:5060:
OPTIONS sip:sipconnect.sipgate.de SIP/2.0
Via: SIP/2.0/UDP 192.168.1.251:5060;branch=z9hG4bK3fad19ce
Max-Forwards: 70
From: "asterisk" <sip:XXXXXXXt0@192.168.1.251>;tag=as314b9aa4
To: <sip:sipconnect.sipgate.de>
Contact: <sip:XXXXXXXt0@192.168.1.251:5060>
Call-ID: 070ee473671582cc67a8cc7c5622ddd3@192.168.1.251:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.6-cert1
Date: Sat, 26 Apr 2014 15:38:32 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
Retransmitting #4 (no NAT) to 217.10.68.150:5060:
OPTIONS sip:sipconnect.sipgate.de SIP/2.0
Via: SIP/2.0/UDP 192.168.1.251:5060;branch=z9hG4bK3fad19ce
Max-Forwards: 70
From: "asterisk" <sip:XXXXXXXt0@192.168.1.251>;tag=as314b9aa4
To: <sip:sipconnect.sipgate.de>
Contact: <sip:XXXXXXXt0@192.168.1.251:5060>
Call-ID: 070ee473671582cc67a8cc7c5622ddd3@192.168.1.251:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.6-cert1
Date: Sat, 26 Apr 2014 15:38:32 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
Really destroying SIP dialog '070ee473671582cc67a8cc7c5622ddd3@192.168.1.251:5060' Method: OPTIONS
mobydick*CLI> sip set debug off

Danach sieht es ja aus, als würde er doch kein NAT benutzen Ein grep nach der externip auf /etc/asterisk/* brachte auch keine Ausgabe. Werden die Einträge aus der Oberfläche evtl. nicht übernommen? Ich habe nach den jeweiligen Änderungen den Telefonieserver neugeladen/neugestartet.

Hat da noch jemand Tipp für mich?

Danke
Ulf

ulf.kosack · April 27, 2014, 2:46am

Hallo zusammen,

manchmal braucht es anscheinend eine Pause. Nachdem ich gestern aufgehört hatte und mich heute mit einem anderen Thema in der MobyDick beschäftigt hatte, funktionieren auf einmal die ausgehenden Verbindungen :D. Eine Erklärung dafür habe ich nicht .

Ich habe zwar noch eine Einstellung geändert, diese aber auch erst nach einem erfolgreichen ausgehenden Verbindungsaufbau und erst als ich im Asterisk Debug Warnings gesehen habe:

  == Parsing '/etc/asterisk/mdc_sip_gw.conf': Found
[Apr 27 11:37:19] WARNING[24838]: chan_sip.c:31817 reload_config: Invalid address for externaddr keyword: dyndns.mydomain.tld
[Apr 27 11:37:19] WARNING[24838]: sip/config_parser.c:812 sip_parse_nat_option: nat=yes is deprecated, use nat=force_rport,comedia instead

Ich habe den Eintrag “nat” entsprechend dem o.g. Hinweis abgeändert und den “Telefonieserver neugeladen” (2x). Dann hat ihn auch nicht mehr der Eintrag für die externaddr gestört.

Ich lasse das Thema vorrübergehend ungelöst, falls doch noch Probleme auftauchen, wie in den o.g. Threads beschrieben, dass nach einer bestimmten Ruhezeit der Fehler wieder auftritt.

Danke
Ulf

volvo1979 · April 6, 2015, 7:10am

Magst Du mal lachen? Ich habe entsprechend auch alle Einstellungen gesetzt, mehrfacher neustart des Systems.

Wenn ich jedoch ein:

sip reload mache, bekomme ich genau den gleichen mist:

Invalid address for externaddr keyword: meine.domain.de

als nächste kommt dann noch ein Fehler:
ERROR[1875]: chan_iax2.c:5096 handle_call_token: Call rejected, CallToken Support required. If unexpected, resolve by placing address 127.0.0.1 in the calltokenoptional list or setting user iaxmodem0 requirecalltoken=no

BTW ich bin Telekomkunde und habe nix mit sipgate zum tun.