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Asterisk (ankommend) funktioniert in sipgate.de aber nicht in sipgate.at !?!?!?

Dieses Thema im Forum "Asterisk Allgemein" wurde erstellt von strubinsky, 23 Feb. 2006.

  1. strubinsky

    strubinsky Neuer User

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    #1 strubinsky, 23 Feb. 2006
    Zuletzt bearbeitet: 23 Feb. 2006
    Meine Tochter lebt in der BRD und Meine Eltern in Österreich. Ich in den USA. Deshalb habe ich mire zwei sipgate accounts besorgt (sipgate.at und sipgate.de)

    Das Misteriöse bei der Sache ist, daß die deutsche nummer super funktioniert, während es beim österreichischen nur ausgehend (von sipgate.at eine Wiener Nummer wählen) klappt. Eingehend sehe ich dass das telefon von asterisk abgehoben wird und die background cmds abgespielt werden. Aber es ist absolut nichts zu hören.

    Ich habe die Musterkonfiguration von hier kopiert und: ja, das klappt. Es läutet und wenn ich abhebe kann ich sprechen (ich rufe meine sipgate nr via POTS von den usa aus an). Aber nix anderes geht. Keine voicemail, keine music on hold, keine background Sprachprompts. All das funktioniert aber wunderbar via sipgate.de.

    Hier der relevante Teil meiner sip.conf:



    [020142637436]
    ;register => 2637436:<pwde>@sipgate.de/020142637436
    type=friend
    context=incoming-voip
    username=2637436
    fromuser=2637436
    authuser=2637436
    disable=all
    allow=alaw
    allow=ulaw
    allow=g729
    allow=gsm
    allow=slinear
    ;dtmfmode=inband ; Choices are inband, rfc2833, or info
    ;dtmf=rfc2833
    insecure=very ; otherwise I get authentication errors
    nat=yes
    fromdomain=sipgate.de
    secret=<pwde>
    host=sipgate.de
    qualify=yes
    ;callerid="guenter strubinsky" <020142637436>
    mailbox=3484@default
    srvlookup=yes
    canreinvite=no

    [19628879]
    type=friend
    ;register => 9628879:<pwat>@sipgate.at/9628879
    context=incoming-voip
    username=9628879
    fromuser=9628879
    authuser=9628879
    disable=all
    allow=alaw
    allow=ulaw
    allow=g729
    allow=gsm
    allow=slinear
    ;dtmfmode=inband ; Choices are inband, rfc2833, or info
    ;dtmf=rfc2833
    insecure=very ; otherwise I get authentication errors
    nat=yes
    fromdomain=sipgate.at
    secret=<pwat>
    host=sipgate.at
    qualify=yes
    ;callerid="guenter strubinsky" <19628879>
    mailbox=3484@default
    srvlookup=yes
    canreinvite=no



    und die extensions.conf:

    ...

    ;----------------------------------------------------------------------------------------
    [macro-stdexten]
    ;----------------------------------------------------------------------------------------
    ;
    ; Standard extension macro:
    ; ${ARG1} - Extension (we could have used ${MACRO_EXTEN} here as well
    ; ${ARG2} - Device(s) to ring
    ;

    exten => s,1,NoOp("stdXt (1) ${ARG1}")
    exten => s,n,NoOp("stdXt (2) ${CALLERIDNUM}")

    exten => s,n,GotoIf($[${ARG1} = ${guenti}]?callGuenti)

    exten => s,n,GotoIf($[${LEN(${CALLERIDNUM})} < 7]?internalCall:externalCall)

    exten => s,n(internalCall),GotoIf($[${LEN(${ARG1})} < 7]?callInside:callOutside)

    exten => s,n(callOutside),NoOp(callOutside)
    exten => s,n,Playtones(ring)
    exten => s,n,SetCallerId("Strubinsky guenter" <402-403-3113>,a)

    exten => s,n,ChanIsAvail(${outLine1})
    exten => s,n,NoOp(${AVAILORIGCHAN})
    exten => s,n,Dial(${AVAILORIGCHAN}/${ARG1},120,${insecwarn})
    ;exten => s,n,Dial(IAX2/0611932438@14028172455/${ARG1},120,A(channel-insecure-warn))
    exten => s,n,Goto(s-${DIALSTATUS},1) ; Jump based on status
    ; (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)

    exten => s,n(callInside),NoOp(callInside)
    exten => s,n,Dial(SIP/${ARG1},20,o)
    exten => s,n,Goto(s-${DIALSTATUS},1) ; Jump based on status
    ; (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)

    exten => s,n(externalCall),NoOp(External Call)
    exten => s,n,Set(LastCalled=${DB(callers/${CALLERIDNUM}/lastcall)})
    exten => s,n,Background(channel-insecure-warn)
    exten => s,n,SetMusicOnHold(default)
    exten => s,n,Dial(SIP/${ARG1},20,om)
    exten => s,n,Goto(s-${DIALSTATUS},1) ; Jump based on status
    ; (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)

    exten => s,n(callGuenti),Macro(guenticall,${ARG1},${ARG2})
    exten => s,n,hangup()

    exten => s-NOANSWER,1,Voicemail(u${ARG1}) ; If unavailable, send to voicemail w/ unavail announce
    exten => s-NOANSWER,2,Goto(dir,411,1) ; If they press #, return to start

    exten => s-BUSY,1,Voicemail(b${ARG1}) ; If busy, send to voicemail w/ busy announce
    exten => s-BUSY,2,Goto(dir,411,1) ; If they press #, return to start

    exten => _s-.,1,Goto(s-NOANSWER,1) ; Treat anything else as no answer

    exten => a,1,VoicemailMain(${ARG1}) ; If they press *, send the user into VoicemailMain

    ;----------------------------------------------------------------------------------------
    [gotExtension]
    ;----------------------------------------------------------------------------------------
    exten => 1,1,NoOp(${CALLERIDNUM}) ; SayNumber(${LEN(${CALLERIDNUM})})
    exten => 1,2,GotoIf($[${LEN(${CALLERIDNUM})} < 7]?1,8)
    exten => 1,n,Background(silence/1) ; Wait a second, just for fun
    exten => 1,n,Set(DB(callers/${CALLERIDNUM}/lastcall)=${toDial})
    exten => 1,n,Background(extension)
    exten => 1,n,SayNumber(${toDial})
    exten => 1,n,Background(transfer)
    exten => 1,n,Macro(stdexten,${toDial},${${dialName}})

    exten => #,1,Voicemail(b${toDial})


    ...

    ;----------------------------------------------------------------------------------------
    [incoming]
    ;----------------------------------------------------------------------------------------
    exten => 1,1,Answer()
    exten => 1,2,NoOp(incoming=(${CALLERIDNUM}))
    exten => 1,3,Background(silence/1)
    exten => 1,4,Set(toDial=411${DB(callers/${CALLERIDNUM}/lastcall)})
    exten => 1,5,GotoIf($[${toDial}=411]?1,wtr0:)
    ;exten => 1,6,Set(toDial=${toDial:3})
    exten => 1,6,Set(toDial=3536)
    exten => 1,7,Background(the-num-i-have-is)
    exten => 1,8,System(/etc/asterisk/filexists /var/spool/asterisk/voicemail/default/${toDial}/greet.gsm) ; test for the existance of this path as a recording
    exten => 1,n,Background(/var/spool/asterisk/voicemail/default/${toDial}/greet)
    exten => 1,n(aftername),Background(to-call-num-press)
    exten => 1,n,Background(pound)
    exten => 1,n,Wait(1)
    exten => 1,n,Goto(repeat)

    exten => 1,109,SayNumber(${toDial})
    exten => 1,n,Goto(aftername)
    sip.conf:

    exten => 1,n(wtr0),Set(toDial=411)
    exten => 1,n(wtr),Background(thanks-for-calling-today)
    exten => 1,n(repeat),Background(if-u-know-ext-dial)
    exten => 1,n,Background(to-dial-by-name)
    exten => 1,n,Background(press-9)
    exten => 1,n,Background(vm-reachoper)
    exten => 1,n,Background(star-for-menu-again)
    exten => 1,n(reentry),NoOp()
    ;----------------------------------------------------------------------------------------
    ;[singleInput]
    ;----------------------------------------------------------------------------------------
    exten => 1,n(singleInput),NoOp(Single input called)
    exten => 1,n,WaitExten()
    exten => 1,n,Background(one-moment-please)
    exten => 1,n,Set(chosen=0)
    ;exten => 1,n,Goto(dir,411,1)
    exten => 1,n,goto(1,repeat)

    exten => i,1,Set(chosen=0)
    exten => i,n,background(i-dont-understand)
    exten => i,n,background(please-try-again)
    exten => i,n,goto(1,repeat)

    exten => _0!,1,NoOp(0)
    exten => _0!,n,Set(chosen=${EXTEN})
    exten => _0!,n,goto(connect,oper) ; hangup

    exten => _#!,1,NoOp(pound)
    exten => _#!,n,Set(chosen=${EXTEN})
    exten => _#!,n,goto(chosen,1) ; hangup

    exten => _9!,1,NoOp(9)
    exten => _9!,n,Set(chosen=${EXTEN})
    exten => _9!,n,goto(dir,411,1) ; hangup

    exten => _*!,1,NoOp(star)
    exten => _*!,n,Set(chosen=${EXTEN})
    exten => _*!,n,goto(1,1) ; hangup

    exten => n,1,sayphonetic(n)
    exten => n,n,Set(chosen=${EXTEN})
    exten => n,n,goto(chosen,1) ; hangup

    exten => a,1,NoOp(a) ; * pressed
    exten => a,n,Set(chosen=${EXTEN})
    exten => a,n,goto(chosen,1) ; hangup

    exten => o,1,NoOp(o) ; operator / 0
    exten => o,n,Set(chosen=${EXTEN})
    exten => o,n,goto(chosen,1) ; hangup

    exten => t,1,NoOp(t) ; Invalid
    exten => t,n,Set(chosen=${EXTEN})
    exten => t,n,goto(chosen,1) ; hangup

    exten => h,1,NoOp(h) ; hangup
    exten => h,n,background(i-dont-understand)
    exten => h,n,background(please-try-again)
    exten => h,n,goto(chosen,1) ; hangup

    exten => s,1,NoOp(s) ; start
    exten => s,n,goto(chosen,1) ; hangup

    exten => _3XXX!,1,Set(toDial=${EXTEN})
    exten => _3XXX!,n,Goto(gotExtension,1,1)

    exten => _8XXX!,1,Goto(default,${EXTEN},1)

    exten => chosen,1,NoOp(${chosen})
    exten => chosen,n,GotoIf(1,$[${chosen}="*"]?1,repeat)
    exten => chosen,n,GotoIf(1,$[${chosen}="9"]?connect,411)
    exten => chosen,n,GotoIf(1,$[${chosen}="0"]?connect,oper)
    exten => chosen,n,GotoIf(1,$[${chosen}="#"]?gotExtension,1,1)
    exten => chosen,n,goto(1,repeat)

    exten => connect,1,NoOp(connect entered)
    exten => connect,n(411),Goto(dir,411,1)
    exten => connect,n(oper),Set(toDial=${oper})
    exten => connect,n,Goto(gotExtension,1,1)

    ;----------------------------------------------------------------------------------------
    [incoming-voip]
    ;----------------------------------------------------------------------------------------

    exten => _1XXXXXXXXXX!,1,Set(LANGUAGE()=en)
    ;exten => _1XXXXXXXXXX!,n,Answer()
    ;exten => _1XXXXXXXXXX!,n,gotoIf($[${EXTEN} = "4022926801"]?susans,1,1)
    exten => _1XXXXXXXXXX!,n,goto(incoming,1,1)
    exten => _X.,1,Set(LANGUAGE()=de)
    ;exten => _X.,n,Answer()
    exten => _X.,n,Background(international-call)
    exten => _X.,n,goto(incoming,1,1)



    wenn es nirgends klappen würde, dann hätte ich einfach ein parr Fehler. Die kann ich ja debuggen.

    ABER;
    Es funktioniert wenn ich
    - lokal (us ortsgespräch - (ankommend/ausgehend),
    - us ferngespräch - (ankommend/ausgehend),
    - brd orts und ferngespräch - (ankommend/ausgehend)
    - at orts und ferngespräch - (AUSGEHEND)

    habe, nur nicht wenn ich AT eingehend habe. Da -wie oben erwähnt- kann ich sehen, wie die Backgroundansagen abgesspielt werden (höre aber nix), die Wartemusik gestarted wird (höre aber nix).

    Wenn wer interessiert ist, kann ich die notwendigen files komplett zippen und attachen. Hat jemand ähnlicher Erfahrungen gemacht?

    Ich habe sogar mit Ethereal die RTP blöcke verfolgt. Alles scheint durchzugehen!!! Die einzige Erklärung die ich habe, ist daß die falschen Codecs für den sound verwendet werden (sipgate anderer als asterisk-sipura spa 2100), aber auch diese habe ich geprüft und es scheint identisch zu sein.

    Ich bin am Ende mit meinem Latein!!!! :noidea: :noidea:
    Kann mir wer helfen?

    günter

    p.s. Um es einfacher zu machen:
    sip.conf:
    [020142637436]
    register => 9628879:<pwat>@sipgate.at/9628879
    type=friend
    ;context=incoming-voip
    context=ankommend
    ...

    extensions.conf
    [ankommend]
    exten => 9628879,1,Answer()
    exten => 9628879,n,Background(international-call)
    exten => 9628879,n,Dial(SIP/3536,20)

    Absolut dasselbe Verhalten(ankommend):
    sipfgate.de :rock:
    sipgate.at :mad:
     
  2. betateilchen

    betateilchen Grandstream-Guru

    Registriert seit:
    30 Juni 2004
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    am Letzenberg
    Dein Problem ist ganze einfach erklärt :D

    Code:
    register => 9628879:<pwat>@sipgate.at/9628879
    register Einträge dürfen niemals in einem anderen Kontext stehen, als in [general] :!: Das dürfte Dein Hauptfehler sein. Du darfst sie nicht in die einzelnen Kontexte der Accounts schreiben. Und "type=friend" ist in aktuellen Asterisk-Versionen auch problematisch.

    Übrigens gibt es hier im Forum auch eine Musterkonfiguration für Sipgate-Accounts auf Asterisk.
     
  3. strubinsky

    strubinsky Neuer User

    Registriert seit:
    22 Feb. 2006
    Beiträge:
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    Tut mir Leid für das Mißverständnis ß-Teilchen; der "Register" Teil ist in [general] . Ich habe es lediglich (als Kommentar) unter der ID stehen.

    Ich dachte, friend ist rein-raus, user ist raus and peer ist rein.

    Okay, ich hab's gefunden, man sollte rein und raus separat behandeln!

    Ich habe es so gemacht, wie Dein Template und bin zu folgendem Resultat gekommen:

    Code:
    [9628879]
    type=user
    username=9628879
    fromuser=9628879
    secret=<pwat>
    host=sipgate.at
    fromdomain=sipgate.at
    insecure=very
    canreinvite=no
    nat=no
    disallow=all
    allow=ulaw
    
    [sipgate_at_in]
    type=peer
    fromdomain=sipgate.at
    ;context=incoming-voip
    host=sipgate.at
    context=ankommend
    
    Version 1:
    Code:
    [ankommend]
    exten => 9628879,1,Answer()
    exten => 9628879,n,Background(international-call)
    exten => 9628879,n,Dial(SIP/3536,20)
    
    Resultat:
    • Ich sehe, daß das telefon beantwortet wird (höre aber nichts am anrufenden Aparat -Schweigen-)
    • Ich sehe, daß die international-call Meldung backgrounded wird (höre aber nichts am anrufenden Aparat -Schweigen-)
    • Ich sehe den Dial Befehl
    • Nebenstelle 3536 läutet
    • ich hebe ab und höre immer noch nichts, (an beiden Seiten) auch wenn ich rede, sowohl am ausgehenden wie ankommenden Apparat.
    • Die Leitung wird mit einem TimeOut automatisch getrennt.

    Protokoll:
    Code:
    <-- SIP read from 8.11.1.5:5060:
    OPTIONS sip:14024033113@68.227.169.158 SIP/2.0
    Via: SIP/2.0/UDP 8.11.1.5:5060;branch=z9hG4bK068525a0;rport
    From: "asterisk" <sip:asterisk@8.11.1.5>;tag=as462e8898
    To: <sip:14024033113@68.227.169.158>
    Contact: <sip:asterisk@8.11.1.5>
    Call-ID: 7e5fd8b576aaed322ec9aab6413607f7@8.11.1.5
    CSeq: 102 OPTIONS
    User-Agent: Viatalk SIP
    Max-Forwards: 70
    Date: Thu, 23 Feb 2006 16:18:18 GMT
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
    Content-Length: 0
    strubinsky*CLI>
    
    --- (12 headers 0 lines)---
    Looking for 14024033113 in default (domain 68.227.169.158)
    Transmitting (no NAT) to 8.11.1.5:5060:<-- SIP read from 217.116.119.252:5060:
    INVITE sip:9628879@68.227.169.158 SIP/2.0
    Record-Route: <sip:9628879@217.116.119.252;ftag=as1b712083;lr=on>
    Max-Forwards:  9
    Record-Route: <sip:4319628879@217.10.79.8;ftag=as1b712083;lr=on>
    Via: SIP/2.0/UDP 217.116.119.252;branch=z9hG4bK548c.55118bd6.1
    Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bK548c.75e12503.0
    Via: SIP/2.0/UDP 217.10.66.71:5060;branch=z9hG4bK78d44b8d
    From: "0414022926801" <sip:0414022926801@217.10.66.71>;tag=as1b712083
    To: <sip:4319628879@sipgate.net>
    Contact: <sip:0414022926801@217.10.66.71>
    Call-ID: 49f8cf4a5a7fb68b6c88a3ae18b7298f@217.10.66.71
    CSeq: 102 INVITE
    User-Agent: sipgate asterisk
    Date: Thu, 23 Feb 2006 16:39:07 GMT
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
    Content-Type: application/sdp
    Content-Length: 411
    
    v=0
    o=root 32613 32613 IN IP4 217.10.66.71
    s=session
    c=IN IP4 217.10.66.71
    t=0 0
    m=audio 12290 RTP/AVP 8 0 3 97 18 2 4 5 110 7 10
    a=rtpmap:8 PCMA/8000
    a=rtpmap:0 PCMU/8000
    a=rtpmap:3 GSM/8000
    a=rtpmap:97 iLBC/8000
    a=rtpmap:18 G729/8000
    a=rtpmap:2 G726-32/8000
    a=rtpmap:4 G723/8000
    a=rtpmap:5 DVI4/8000
    a=rtpmap:110 speex/8000
    a=rtpmap:7 LPC/8000
    a=rtpmap:10 L16/8000
    a=silenceSupp:off - - - -
    
    --- (17 headers 18 lines)---
    Using INVITE request as basis request - 49f8cf4a5a7fb68b6c88a3ae18b7298f@217.10.66.71
    Sending to 217.116.119.252 : 5060 (non-NAT)
    Found peer 'sipgate_at_in'
    Found RTP audio format 8
    Found RTP audio format 0
    Found RTP audio format 3
    Found RTP audio format 97
    Found RTP audio format 18
    Found RTP audio format 2
    Found RTP audio format 4
    Found RTP audio format 5
    Found RTP audio format 110
    Found RTP audio format 7
    Found RTP audio format 10
    Peer audio RTP is at port 217.10.66.71:12290
    Found description format PCMA
    Found description format PCMU
    Found description format GSM
    Found description format iLBC
    Found description format G729
    Found description format G726-32
    Found description format G723
    Found description format DVI4
    Found description format speex
    Found description format LPC
    Found description format L16
    Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x7ff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
    Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
    Looking for 9628879 in ankommend (domain 68.227.169.158)
    list_route: hop: <sip:9628879@217.116.119.252;ftag=as1b712083;lr=on>
    list_route: hop: <sip:4319628879@217.10.79.8;ftag=as1b712083;lr=on>
    Transmitting (no NAT) to 217.116.119.252:5060:
    SIP/2.0 100 Trying
    Via: SIP/2.0/UDP 217.116.119.252;branch=z9hG4bK548c.55118bd6.1;received=217.116.119.252
    Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bK548c.75e12503.0
    Via: SIP/2.0/UDP 217.10.66.71:5060;branch=z9hG4bK78d44b8d
    From: "0414022926801" <sip:0414022926801@217.10.66.71>;tag=as1b712083
    To: <sip:4319628879@sipgate.net>
    Call-ID: 49f8cf4a5a7fb68b6c88a3ae18b7298f@217.10.66.71
    CSeq: 102 INVITE
    User-Agent: Asterisk PBX
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
    Contact: <sip:9628879@68.227.169.158>
    Content-Length: 0
    
    
    ---
        -- Executing Set("SIP/217.10.66.71-09d10448", "apparat=3536") in new stack
        -- Executing Dial("SIP/217.10.66.71-09d10448", "SIP/3536|20") in new stack
    We're at 192.168.11.69 port 18882
    Adding codec 0x4 (ulaw) to SDP
    Adding codec 0x2 (gsm) to SDP
    Adding codec 0x8 (alaw) to SDP
    Adding codec 0x10 (g726) to SDP
    Adding codec 0x100 (g729) to SDP
    Adding codec 0x400 (ilbc) to SDP
    Adding non-codec 0x1 (telephone-event) to SDP
    14 headers, 16 lines
    Reliably Transmitting (no NAT) to 192.168.11.202:5060:
    INVITE sip:3536@192.168.11.202:5060 SIP/2.0
    Via: SIP/2.0/UDP 192.168.11.69:5060;branch=z9hG4bK2da50bf2;rport
    From: "0414022926801" <sip:0414022926801@192.168.11.69>;tag=as51f3487a
    To: <sip:3536@192.168.11.202:5060>
    Contact: <sip:0414022926801@192.168.11.69>
    Call-ID: 1111d2c7243d4e4551fef97d65167ed9@192.168.11.69
    CSeq: 102 INVITE
    User-Agent: Asterisk PBX
    Max-Forwards: 70
    Remote-Party-ID: "0414022926801" <sip:0414022926801@192.168.11.69>;privacy=off;screen=no
    Date: Thu, 23 Feb 2006 16:39:08 GMT
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
    Content-Type: application/sdp
    Content-Length: 367
    
    v=0
    o=root 2558 2558 IN IP4 192.168.11.69
    s=session
    c=IN IP4 192.168.11.69
    t=0 0
    m=audio 18882 RTP/AVP 0 3 8 111 18 97 101
    a=rtpmap:0 PCMU/8000
    a=rtpmap:3 GSM/8000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:111 G726-32/8000
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=no
    a=rtpmap:97 iLBC/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=silenceSupp:off - - - -
    
    ---
        -- Called 3536
    strubinsky*CLI>
    <-- SIP read from 192.168.11.202:5060:
    SIP/2.0 100 Trying
    To: <sip:3536@192.168.11.202:5060>
    From: "0414022926801" <sip:0414022926801@192.168.11.69>;tag=as51f3487a
    Call-ID: 1111d2c7243d4e4551fef97d65167ed9@192.168.11.69
    CSeq: 102 INVITE
    Via: SIP/2.0/UDP 192.168.11.69:5060;branch=z9hG4bK2da50bf2
    Server: Sipura/SPA2100-2.0.5(d)
    Content-Length: 0
    
    
    --- (8 headers 0 lines)---
    
    <-- SIP read from 192.168.11.202:5060:
    SIP/2.0 180 Ringing
    To: <sip:3536@192.168.11.202:5060>;tag=ba310c92db31bcc1i0
    rom: "0414022926801" <sip:0414022926801@192.168.11.69>;tag=as51f3487a
    Call-ID: 1111d2c7243d4e4551fef97d65167ed9@192.168.11.69
    CSeq: 102 INVITE
    Via: SIP/2.0/UDP 192.168.11.69:5060;branch=z9hG4bK2da50bf2
    Server: Sipura/SPA2100-2.0.5(d)
    Content-Length: 0
    
    
    --- (8 headers 0 lines)---
        -- SIP/3536-443b is ringing
    Transmitting (no NAT) to 217.116.119.252:5060:
    SIP/2.0 180 Ringing
    Via: SIP/2.0/UDP 217.116.119.252;branch=z9hG4bK548c.55118bd6.1;received=217.116.119.252
    Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bK548c.75e12503.0
    Via: SIP/2.0/UDP 217.10.66.71:5060;branch=z9hG4bK78d44b8d
    From: "0414022926801" <sip:0414022926801@217.10.66.71>;tag=as1b712083
    To: <sip:4319628879@sipgate.net>;tag=as218b8918
    Call-ID: 49f8cf4a5a7fb68b6c88a3ae18b7298f@217.10.66.71
    CSeq: 102 INVITE
    User-Agent: Asterisk PBX
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
    Contact: <sip:9628879@68.227.169.158>
    Content-Length: 0
    
    
    ---
    strubinsky*CLI>
    <-- SIP read from 192.168.11.202:5060:
    SIP/2.0 200 OK
    To: <sip:3536@192.168.11.202:5060>;tag=ba310c92db31bcc1i0
    From: "0414022926801" <sip:0414022926801@192.168.11.69>;tag=as51f3487a
    Call-ID: 1111d2c7243d4e4551fef97d65167ed9@192.168.11.69
    CSeq: 102 INVITE
    Via: SIP/2.0/UDP 192.168.11.69:5060;branch=z9hG4bK2da50bf2
    Contact: 3536 <sip:3536@192.168.11.202:5060>
    Server: Sipura/SPA2100-2.0.5(d)
    Content-Length: 241
    Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
    Supported: x-sipura
    Content-Type: application/sdp
    
    v=0
    o=- 19758508 19758508 IN IP4 192.168.11.202
    s=-
    c=IN IP4 192.168.11.202
    t=0 0
    m=audio 16386 RTP/AVP 0 100 101
    a=rtpmap:0 PCMU/8000
    a=rtpmap:100 NSE/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    a=ptime:30
    a=sendrecv
    
    --- (12 headers 12 lines)---
    Found RTP audio format 0
    Found RTP audio format 100
    Found RTP audio format 101
    Peer audio RTP is at port 192.168.11.202:16386
    Found description format PCMU
    Found description format NSE
    Found description format telephone-event
    Capabilities: us - 0x51e (gsm|ulaw|alaw|g726|g729|ilbc), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
    Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
    list_route: hop: <sip:3536@192.168.11.202:5060>
    set_destination: Parsing <sip:3536@192.168.11.202:5060> for address/port to send to
    set_destination: set destination to 192.168.11.202, port 5060
    Transmitting (no NAT) to 192.168.11.202:5060:
    ACK sip:3536@192.168.11.202:5060 SIP/2.0
    Via: SIP/2.0/UDP 192.168.11.69:5060;branch=z9hG4bK50c9bee7;rport
    From: "0414022926801" <sip:0414022926801@192.168.11.69>;tag=as51f3487a
    To: <sip:3536@192.168.11.202:5060>;tag=ba310c92db31bcc1i0
    Contact: <sip:0414022926801@192.168.11.69>
    Call-ID: 1111d2c7243d4e4551fef97d65167ed9@192.168.11.69
    CSeq: 102 ACK
    User-Agent: Asterisk PBX
    Max-Forwards: 70
    Remote-Party-ID: "0414022926801" <sip:0414022926801@192.168.11.69>;privacy=off;screen=no
    Content-Length: 0
    
    
    ---
        -- SIP/3536-443b answered SIP/217.10.66.71-09d10448
    We're at 68.227.169.158 port 15630
    Adding codec 0x2 (gsm) to SDP
    Adding codec 0x4 (ulaw) to SDP
    Adding codec 0x8 (alaw) to SDP
    Reliably Transmitting (no NAT) to 217.116.119.252:5060:
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 217.116.119.252;branch=z9hG4bK548c.55118bd6.1;received=217.116.119.252
    Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bK548c.75e12503.0
    Via: SIP/2.0/UDP 217.10.66.71:5060;branch=z9hG4bK78d44b8d
    Record-Route: <sip:9628879@217.116.119.252;ftag=as1b712083;lr=on>
    Record-Route: <sip:4319628879@217.10.79.8;ftag=as1b712083;lr=on>
    From: "0414022926801" <sip:0414022926801@217.10.66.71>;tag=as1b712083
    To: <sip:4319628879@sipgate.net>;tag=as218b8918
    Call-ID: 49f8cf4a5a7fb68b6c88a3ae18b7298f@217.10.66.71
    CSeq: 102 INVITE
    User-Agent: Asterisk PBX
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
    Contact: <sip:9628879@68.227.169.158>
    Content-Type: application/sdp
    Content-Length: 209
    
    v=0
    o=root 2558 2558 IN IP4 68.227.169.158
    s=session
    c=IN IP4 68.227.169.158
    t=0 0
    m=audio 15630 RTP/AVP 3 0 8
    a=rtpmap:3 GSM/8000
    a=rtpmap:0 PCMU/8000
    a=rtpmap:8 PCMA/8000
    a=silenceSupp:off - - - -
    
    ---
        -- Attempting native bridge of SIP/217.10.66.71-09d10448 and SIP/3536-443b
    strubinsky*CLI>
    <-- SIP read from 217.116.119.252:5060:
    ACK sip:9628879@68.227.169.158 SIP/2.0
    Record-Route: <sip:9628879@217.116.119.252;ftag=as1b712083;lr=on>
    Max-Forwards:  9
    Via: SIP/2.0/UDP 217.116.119.252;branch=0
    Via: SIP/2.0/UDP 217.10.79.8;branch=0
    Via: SIP/2.0/UDP 217.10.66.71:5060;branch=z9hG4bK11696cd9
    From: "0414022926801" <sip:0414022926801@217.10.66.71>;tag=as1b712083
    To: <sip:4319628879@sipgate.net>;tag=as218b8918
    Contact: <sip:0414022926801@217.10.66.71>
    Call-ID: 49f8cf4a5a7fb68b6c88a3ae18b7298f@217.10.66.71
    CSeq: 102 ACK
    User-Agent: sipgate asterisk
    Content-Length: 0
    
    
    --- (13 headers 0 lines)---
    Destroying call '4113057f50ac696d2edd114f574254a9@127.0.0.1'
    Destroying call '66010d976234e4c72ffc4eb07534a508@127.0.0.1'
    Destroying call '2974ff062b7dd5684ec0512a27157685@127.0.0.1'
    strubinsky*CLI>
    <-- SIP read from 192.168.11.202:5060:
    BYE sip:0414022926801@192.168.11.69 SIP/2.0
    Via: SIP/2.0/UDP 192.168.11.202:5060;branch=z9hG4bK-cf15f509
    From: <sip:3536@192.168.11.202:5060>;tag=ba310c92db31bcc1i0
    To: "0414022926801" <sip:0414022926801@192.168.11.69>;tag=as51f3487a
    Call-ID: 1111d2c7243d4e4551fef97d65167ed9@192.168.11.69
    CSeq: 101 BYE
    Max-Forwards: 70
    User-Agent: Sipura/SPA2100-2.0.5(d)
    Content-Length: 0
    
    SIP/2.0 404 Not Found
    Via: SIP/2.0/UDP 8.11.1.5:5060;branch=z9hG4bK068525a0;rport;received=8.11.1.5
    From: "asterisk" <sip:asterisk@8.11.1.5>;tag=as462e8898
    To: <sip:14024033113@68.227.169.158>;tag=as39e1815e
    Call-ID: 7e5fd8b576aaed322ec9aab6413607f7@8.11.1.5
    CSeq: 102 OPTIONS
    User-Agent: Asterisk PBX
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
    Contact: <sip:68.227.169.158>
    Accept: application/sdp
    Content-Length: 0
    
    
    ---
    Destroying call '7e5fd8b576aaed322ec9aab6413607f7@8.11.1.5'
    strubinsky*CLI>
    <-- SIP read from 217.116.119.252:5060:
    INVITE sip:9628879@68.227.169.158 SIP/2.0
    Record-Route: <sip:9628879@217.116.119.252;ftag=as2590f98a;lr=on>
    Max-Forwards:  9
    Record-Route: <sip:4319628879@217.10.79.8;ftag=as2590f98a;lr=on>
    Via: SIP/2.0/UDP 217.116.119.252;branch=z9hG4bK84d8.8e344746.1
    Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bK84d8.e9167642.0
    Via: SIP/2.0/UDP 217.10.66.71:5060;branch=z9hG4bK1d285581
    From: "043" <sip:043@217.10.66.71>;tag=as2590f98a
    To: <sip:4319628879@sipgate.net>
    Contact: <sip:043@217.10.66.71>
    Call-ID: 2da8e58620d42242408a8a6648b7e42e@217.10.66.71
    CSeq: 102 INVITE
    User-Agent: sipgate asterisk
    Date: Thu, 23 Feb 2006 16:31:13 GMT
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
    Content-Type: application/sdp
    Content-Length: 411
    
    v=0
    o=root 31839 31839 IN IP4 217.10.66.71
    s=session
    c=IN IP4 217.10.66.71
    t=0 0
    m=audio 10408 RTP/AVP 8 0 3 97 18 2 4 5 110 7 10
    a=rtpmap:8 PCMA/8000
    a=rtpmap:0 PCMU/8000
    a=rtpmap:3 GSM/8000
    a=rtpmap:97 iLBC/8000
    a=rtpmap:18 G729/8000
    a=rtpmap:2 G726-32/8000
    a=rtpmap:4 G723/8000
    a=rtpmap:5 DVI4/8000
    a=rtpmap:110 speex/8000
    a=rtpmap:7 LPC/8000
    a=rtpmap:10 L16/8000
    a=silenceSupp:off - - - -
    
    --- (17 headers 18 lines)---
    Using INVITE request as basis request - 2da8e58620d42242408a8a6648b7e42e@217.10.66.71
    Sending to 217.116.119.252 : 5060 (non-NAT)
    Found peer 'sipgate_at_in'
    Found RTP audio format 8
    Found RTP audio format 0
    Found RTP audio format 3
    Found RTP audio format 97
    Found RTP audio format 18
    Found RTP audio format 2
    Found RTP audio format 4
    Found RTP audio format 5
    Found RTP audio format 110
    Found RTP audio format 7
    Found RTP audio format 10
    Peer audio RTP is at port 217.10.66.71:10408
    Found description format PCMA
    Found description format PCMU
    Found description format GSM
    Found description format iLBC
    Found description format G729
    Found description format G726-32
    Found description format G723
    Found description format DVI4
    Found description format speex
    Found description format LPC
    Found description format L16
    Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x7ff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
    Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
    Looking for 9628879 in ankommend (domain 68.227.169.158)
    list_route: hop: <sip:9628879@217.116.119.252;ftag=as2590f98a;lr=on>
    list_route: hop: <sip:4319628879@217.10.79.8;ftag=as2590f98a;lr=on>
    Transmitting (no NAT) to 217.116.119.252:5060:
    SIP/2.0 100 Trying
    Via: SIP/2.0/UDP 217.116.119.252;branch=z9hG4bK84d8.8e344746.1;received=217.116.119.252
    Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bK84d8.e9167642.0
    Via: SIP/2.0/UDP 217.10.66.71:5060;branch=z9hG4bK1d285581
    From: "043" <sip:043@217.10.66.71>;tag=as2590f98a
    To: <sip:4319628879@sipgate.net>
    Call-ID: 2da8e58620d42242408a8a6648b7e42e@217.10.66.71
    CSeq: 102 INVITE
    User-Agent: Asterisk PBX
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
    Contact: <sip:9628879@68.227.169.158>
    Content-Length: 0
    
    
    ---
        -- Executing Answer("SIP/217.10.66.71-09d159b8", "") in new stack
    We're at 68.227.169.158 port 18038
    Adding codec 0x2 (gsm) to SDP
    Adding codec 0x4 (ulaw) to SDP
    Adding codec 0x8 (alaw) to SDP
    Reliably Transmitting (no NAT) to 217.116.119.252:5060:
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 217.116.119.252;branch=z9hG4bK84d8.8e344746.1;received=217.116.119.252
    Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bK84d8.e9167642.0
    Via: SIP/2.0/UDP 217.10.66.71:5060;branch=z9hG4bK1d285581
    Record-Route: <sip:9628879@217.116.119.252;ftag=as2590f98a;lr=on>
    Record-Route: <sip:4319628879@217.10.79.8;ftag=as2590f98a;lr=on>
    From: "043" <sip:043@217.10.66.71>;tag=as2590f98a
    To: <sip:4319628879@sipgate.net>;tag=as281a85bc
    Call-ID: 2da8e58620d42242408a8a6648b7e42e@217.10.66.71
    CSeq: 102 INVITE
    User-Agent: Asterisk PBX
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
    Contact: <sip:9628879@68.227.169.158>
    Content-Type: application/sdp
    Content-Length: 209
    
    v=0
    o=root 2558 2558 IN IP4 68.227.169.158
    s=session
    c=IN IP4 68.227.169.158
    t=0 0
    m=audio 18038 RTP/AVP 3 0 8
    a=rtpmap:3 GSM/8000
    a=rtpmap:0 PCMU/8000
    a=rtpmap:8 PCMA/8000
    a=silenceSupp:off - - - -
    
    ---
    Feb 23 10:31:14 WARNING[12593]: channel.c:787 channel_find_locked: Avoided initial deadlock for '0x9d00bf0', 10 retries!
        -- Executing BackGround("SIP/217.10.66.71-09d159b8", "international-call") in new stack
        -- Playing 'international-call' (language 'en')
    strubinsky*CLI>
    <-- SIP read from 217.116.119.252:5060:
    ACK sip:9628879@68.227.169.158 SIP/2.0
    Record-Route: <sip:9628879@217.116.119.252;ftag=as2590f98a;lr=on>
    Max-Forwards:  9
    Via: SIP/2.0/UDP 217.116.119.252;branch=0
    Via: SIP/2.0/UDP 217.10.79.8;branch=0
    Via: SIP/2.0/UDP 217.10.66.71:5060;branch=z9hG4bK135ce12e
    From: "043" <sip:043@217.10.66.71>;tag=as2590f98a
    To: <sip:4319628879@sipgate.net>;tag=as281a85bc
    Contact: <sip:043@217.10.66.71>
    Call-ID: 2da8e58620d42242408a8a6648b7e42e@217.10.66.71
    CSeq: 102 ACK
    User-Agent: sipgate asterisk
    Content-Length: 0
    
    
    --- (13 headers 0 lines)---
        -- Executing Dial("SIP/217.10.66.71-09d159b8", "SIP/3536|20") in new stack
    We're at 192.168.11.69 port 10884
    Adding codec 0x4 (ulaw) to SDP
    Adding codec 0x2 (gsm) to SDP
    Adding codec 0x8 (alaw) to SDP
    Adding codec 0x10 (g726) to SDP
    Adding codec 0x100 (g729) to SDP
    Adding codec 0x400 (ilbc) to SDP
    Adding non-codec 0x1 (telephone-event) to SDP
    14 headers, 16 lines
    Reliably Transmitting (no NAT) to 192.168.11.202:5060:
    INVITE sip:3536@192.168.11.202:5060 SIP/2.0
    Via: SIP/2.0/UDP 192.168.11.69:5060;branch=z9hG4bK4d9a92cd;rport
    From: "043" <sip:043@192.168.11.69>;tag=as298b4c04
    To: <sip:3536@192.168.11.202:5060>
    Contact: <sip:043@192.168.11.69>
    Call-ID: 255308cd17fc21a7175ac2bf5dbee28f@192.168.11.69
    CSeq: 102 INVITE
    User-Agent: Asterisk PBX
    Max-Forwards: 70
    Remote-Party-ID: "043" <sip:043@192.168.11.69>;privacy=off;screen=no
    Date: Thu, 23 Feb 2006 16:31:16 GMT
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
    Content-Type: application/sdp
    Content-Length: 367
    
    v=0
    o=root 2558 2558 IN IP4 192.168.11.69
    s=session
    c=IN IP4 192.168.11.69
    t=0 0
    m=audio 10884 RTP/AVP 0 3 8 111 18 97 101
    a=rtpmap:0 PCMU/8000
    a=rtpmap:3 GSM/8000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:111 G726-32/8000
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=no
    a=rtpmap:97 iLBC/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=silenceSupp:off - - - -
    
    ---
        -- Called 3536
    strubinsky*CLI>
    <-- SIP read from 192.168.11.202:5060:
    SIP/2.0 100 Trying
    To: <sip:3536@192.168.11.202:5060>
    From: "043" <sip:043@192.168.11.69>;tag=as298b4c04
    Call-ID: 255308cd17fc21a7175ac2bf5dbee28f@192.168.11.69
    CSeq: 102 INVITE
    Via: SIP/2.0/UDP 192.168.11.69:5060;branch=z9hG4bK4d9a92cd
    Server: Sipura/SPA2100-2.0.5(d)
    Content-Length: 0
    
    
    --- (8 headers 0 lines)---
    
    <-- SIP read from 192.168.11.202:5060:
    SIP/2.0 180 Ringing
    To: <sip:3536@192.168.11.202:5060>;tag=fab189c59911f0e4i0
    From: "043" <sip:043@192.168.11.69>;tag=as298b4c04
    Call-ID: 255308cd17fc21a7175ac2bf5dbee28f@192.168.11.69
    CSeq: 102 INVITE
    Via: SIP/2.0/UDP 192.168.11.69:5060;branch=z9hG4bK4d9a92cd
    Server: Sipura/SPA2100-2.0.5(d)
    Content-Length: 0
    
    
    --- (8 headers 0 lines)---
        -- SIP/3536-b3bd is ringing
    Destroying call '057eb09f0a0bcab86d3cfa1e0b104149@127.0.0.1'
    Destroying call '781a4de24a5492cb03d257701fb6570d@127.0.0.1'
    Destroying call '468ca73c2870be8775bb539f529a9c4e@127.0.0.1'
    Destroying call '45054ab6148129677b3028da3ffd2eb2@127.0.0.1'
    Destroying call '511af145253f111849dd8fb77b56ccc8@127.0.0.1'
    Destroying call '1a8d5efa6e78a07947e115f844e99ba2@127.0.0.1'
    strubinsky*CLI>
    
    ANDERS hingegen:
    Code:
    [ankommend]
    ;exten => 9628879,1,Answer()
    ;exten => 9628879,n,Background(international-call)
    exten => 9628879,1,Set(apparat=3536)
    exten => 9628879,n,Dial(SIP/${apparat},20)
    exten => 9628879,n,Goto(1-${DIALSTATUS},1)			; Jump based on status 
    ;									  (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
    
    exten => 1-NOANSWER,1,Voicemail(u${${apparat}})		; If unavailable, send to voicemail w/ unavail announce
    
    Das Telefon läutet (im Gegensatz zu oben wo nichtsa zu hören ist kann ich am anrufenden Apparat das Freizeichen hören), ich kann auf beiden Seiten sprechen und hören. Wenn ich nicht abhebe dann kickt NICHT meine lokale Voicemail ein sondern die von Sipgate.

    Code:
    Bitte beachte das ALLES in Sipgate.de funktioniert !!!!!! 
    
    Protokoll (bei abheben):
    Code:
    <-- SIP read from 217.116.119.252:5060:
    INVITE sip:9628879@68.227.169.158 SIP/2.0
    Record-Route: <sip:9628879@217.116.119.252;ftag=as1b712083;lr=on>
    Max-Forwards:  9
    Record-Route: <sip:4319628879@217.10.79.8;ftag=as1b712083;lr=on>
    Via: SIP/2.0/UDP 217.116.119.252;branch=z9hG4bK548c.55118bd6.1
    Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bK548c.75e12503.0
    Via: SIP/2.0/UDP 217.10.66.71:5060;branch=z9hG4bK78d44b8d
    From: "0414022926801" <sip:0414022926801@217.10.66.71>;tag=as1b712083
    To: <sip:4319628879@sipgate.net>
    Contact: <sip:0414022926801@217.10.66.71>
    Call-ID: 49f8cf4a5a7fb68b6c88a3ae18b7298f@217.10.66.71
    CSeq: 102 INVITE
    User-Agent: sipgate asterisk
    Date: Thu, 23 Feb 2006 16:39:07 GMT
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
    Content-Type: application/sdp
    Content-Length: 411
    
    v=0
    o=root 32613 32613 IN IP4 217.10.66.71
    s=session
    c=IN IP4 217.10.66.71
    t=0 0
    m=audio 12290 RTP/AVP 8 0 3 97 18 2 4 5 110 7 10
    a=rtpmap:8 PCMA/8000
    a=rtpmap:0 PCMU/8000
    a=rtpmap:3 GSM/8000
    a=rtpmap:97 iLBC/8000
    a=rtpmap:18 G729/8000
    a=rtpmap:2 G726-32/8000
    a=rtpmap:4 G723/8000
    a=rtpmap:5 DVI4/8000
    a=rtpmap:110 speex/8000
    a=rtpmap:7 LPC/8000
    a=rtpmap:10 L16/8000
    a=silenceSupp:off - - - -
    
    --- (17 headers 18 lines)---
    Using INVITE request as basis request - 49f8cf4a5a7fb68b6c88a3ae18b7298f@217.10.66.71
    Sending to 217.116.119.252 : 5060 (non-NAT)
    Found peer 'sipgate_at_in'
    Found RTP audio format 8
    Found RTP audio format 0
    Found RTP audio format 3
    Found RTP audio format 97
    Found RTP audio format 18
    Found RTP audio format 2
    Found RTP audio format 4
    Found RTP audio format 5
    Found RTP audio format 110
    Found RTP audio format 7
    Found RTP audio format 10
    Peer audio RTP is at port 217.10.66.71:12290
    Found description format PCMA
    Found description format PCMU
    Found description format GSM
    Found description format iLBC
    Found description format G729
    Found description format G726-32
    Found description format G723
    Found description format DVI4
    Found description format speex
    Found description format LPC
    Found description format L16
    Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x7ff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
    Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
    Looking for 9628879 in ankommend (domain 68.227.169.158)
    list_route: hop: <sip:9628879@217.116.119.252;ftag=as1b712083;lr=on>
    list_route: hop: <sip:4319628879@217.10.79.8;ftag=as1b712083;lr=on>
    Transmitting (no NAT) to 217.116.119.252:5060:
    SIP/2.0 100 Trying
    Via: SIP/2.0/UDP 217.116.119.252;branch=z9hG4bK548c.55118bd6.1;received=217.116.119.252
    Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bK548c.75e12503.0
    Via: SIP/2.0/UDP 217.10.66.71:5060;branch=z9hG4bK78d44b8d
    From: "0414022926801" <sip:0414022926801@217.10.66.71>;tag=as1b712083
    To: <sip:4319628879@sipgate.net>
    Call-ID: 49f8cf4a5a7fb68b6c88a3ae18b7298f@217.10.66.71
    CSeq: 102 INVITE
    User-Agent: Asterisk PBX
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
    Contact: <sip:9628879@68.227.169.158>
    Content-Length: 0
    
    
    ---
        -- Executing Set("SIP/217.10.66.71-09d10448", "apparat=3536") in new stack
        -- Executing Dial("SIP/217.10.66.71-09d10448", "SIP/3536|20") in new stack
    We're at 192.168.11.69 port 18882
    Adding codec 0x4 (ulaw) to SDP
    Adding codec 0x2 (gsm) to SDP
    Adding codec 0x8 (alaw) to SDP
    Adding codec 0x10 (g726) to SDP
    Adding codec 0x100 (g729) to SDP
    Adding codec 0x400 (ilbc) to SDP
    Adding non-codec 0x1 (telephone-event) to SDP
    14 headers, 16 lines
    Reliably Transmitting (no NAT) to 192.168.11.202:5060:
    INVITE sip:3536@192.168.11.202:5060 SIP/2.0
    Via: SIP/2.0/UDP 192.168.11.69:5060;branch=z9hG4bK2da50bf2;rport
    From: "0414022926801" <sip:0414022926801@192.168.11.69>;tag=as51f3487a
    To: <sip:3536@192.168.11.202:5060>
    Contact: <sip:0414022926801@192.168.11.69>
    Call-ID: 1111d2c7243d4e4551fef97d65167ed9@192.168.11.69
    CSeq: 102 INVITE
    User-Agent: Asterisk PBX
    Max-Forwards: 70
    Remote-Party-ID: "0414022926801" <sip:0414022926801@192.168.11.69>;privacy=off;screen=no
    Date: Thu, 23 Feb 2006 16:39:08 GMT
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
    Content-Type: application/sdp
    Content-Length: 367
    
    v=0
    o=root 2558 2558 IN IP4 192.168.11.69
    s=session
    c=IN IP4 192.168.11.69
    t=0 0
    m=audio 18882 RTP/AVP 0 3 8 111 18 97 101
    a=rtpmap:0 PCMU/8000
    a=rtpmap:3 GSM/8000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:111 G726-32/8000
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=no
    a=rtpmap:97 iLBC/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=silenceSupp:off - - - -
    
    ---
        -- Called 3536
    strubinsky*CLI>
    <-- SIP read from 192.168.11.202:5060:
    SIP/2.0 100 Trying
    To: <sip:3536@192.168.11.202:5060>
    From: "0414022926801" <sip:0414022926801@192.168.11.69>;tag=as51f3487a
    Call-ID: 1111d2c7243d4e4551fef97d65167ed9@192.168.11.69
    CSeq: 102 INVITE
    Via: SIP/2.0/UDP 192.168.11.69:5060;branch=z9hG4bK2da50bf2
    Server: Sipura/SPA2100-2.0.5(d)
    Content-Length: 0
    
    
    --- (8 headers 0 lines)---
    
    <-- SIP read from 192.168.11.202:5060:
    SIP/2.0 180 Ringing
    To: <sip:3536@192.168.11.202:5060>;tag=ba310c92db31bcc1i0
    rom: "0414022926801" <sip:0414022926801@192.168.11.69>;tag=as51f3487a
    Call-ID: 1111d2c7243d4e4551fef97d65167ed9@192.168.11.69
    CSeq: 102 INVITE
    Via: SIP/2.0/UDP 192.168.11.69:5060;branch=z9hG4bK2da50bf2
    Server: Sipura/SPA2100-2.0.5(d)
    Content-Length: 0
    
    
    --- (8 headers 0 lines)---
        -- SIP/3536-443b is ringing
    Transmitting (no NAT) to 217.116.119.252:5060:
    SIP/2.0 180 Ringing
    Via: SIP/2.0/UDP 217.116.119.252;branch=z9hG4bK548c.55118bd6.1;received=217.116.119.252
    Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bK548c.75e12503.0
    Via: SIP/2.0/UDP 217.10.66.71:5060;branch=z9hG4bK78d44b8d
    From: "0414022926801" <sip:0414022926801@217.10.66.71>;tag=as1b712083
    To: <sip:4319628879@sipgate.net>;tag=as218b8918
    Call-ID: 49f8cf4a5a7fb68b6c88a3ae18b7298f@217.10.66.71
    CSeq: 102 INVITE
    User-Agent: Asterisk PBX
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
    Contact: <sip:9628879@68.227.169.158>
    Content-Length: 0
    
    
    ---
    strubinsky*CLI>
    <-- SIP read from 192.168.11.202:5060:
    SIP/2.0 200 OK
    To: <sip:3536@192.168.11.202:5060>;tag=ba310c92db31bcc1i0
    From: "0414022926801" <sip:0414022926801@192.168.11.69>;tag=as51f3487a
    Call-ID: 1111d2c7243d4e4551fef97d65167ed9@192.168.11.69
    CSeq: 102 INVITE
    Via: SIP/2.0/UDP 192.168.11.69:5060;branch=z9hG4bK2da50bf2
    Contact: 3536 <sip:3536@192.168.11.202:5060>
    Server: Sipura/SPA2100-2.0.5(d)
    Content-Length: 241
    Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
    Supported: x-sipura
    Content-Type: application/sdp
    
    v=0
    o=- 19758508 19758508 IN IP4 192.168.11.202
    s=-
    c=IN IP4 192.168.11.202
    t=0 0
    m=audio 16386 RTP/AVP 0 100 101
    a=rtpmap:0 PCMU/8000
    a=rtpmap:100 NSE/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    a=ptime:30
    a=sendrecv
    
    --- (12 headers 12 lines)---
    Found RTP audio format 0
    Found RTP audio format 100
    Found RTP audio format 101
    Peer audio RTP is at port 192.168.11.202:16386
    Found description format PCMU
    Found description format NSE
    Found description format telephone-event
    Capabilities: us - 0x51e (gsm|ulaw|alaw|g726|g729|ilbc), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
    Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
    list_route: hop: <sip:3536@192.168.11.202:5060>
    set_destination: Parsing <sip:3536@192.168.11.202:5060> for address/port to send to
    set_destination: set destination to 192.168.11.202, port 5060
    Transmitting (no NAT) to 192.168.11.202:5060:
    ACK sip:3536@192.168.11.202:5060 SIP/2.0
    Via: SIP/2.0/UDP 192.168.11.69:5060;branch=z9hG4bK50c9bee7;rport
    From: "0414022926801" <sip:0414022926801@192.168.11.69>;tag=as51f3487a
    To: <sip:3536@192.168.11.202:5060>;tag=ba310c92db31bcc1i0
    Contact: <sip:0414022926801@192.168.11.69>
    Call-ID: 1111d2c7243d4e4551fef97d65167ed9@192.168.11.69
    CSeq: 102 ACK
    User-Agent: Asterisk PBX
    Max-Forwards: 70
    Remote-Party-ID: "0414022926801" <sip:0414022926801@192.168.11.69>;privacy=off;screen=no
    Content-Length: 0
    
    
    ---
        -- SIP/3536-443b answered SIP/217.10.66.71-09d10448
    We're at 68.227.169.158 port 15630
    Adding codec 0x2 (gsm) to SDP
    Adding codec 0x4 (ulaw) to SDP
    Adding codec 0x8 (alaw) to SDP
    Reliably Transmitting (no NAT) to 217.116.119.252:5060:
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 217.116.119.252;branch=z9hG4bK548c.55118bd6.1;received=217.116.119.252
    Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bK548c.75e12503.0
    Via: SIP/2.0/UDP 217.10.66.71:5060;branch=z9hG4bK78d44b8d
    Record-Route: <sip:9628879@217.116.119.252;ftag=as1b712083;lr=on>
    Record-Route: <sip:4319628879@217.10.79.8;ftag=as1b712083;lr=on>
    From: "0414022926801" <sip:0414022926801@217.10.66.71>;tag=as1b712083
    To: <sip:4319628879@sipgate.net>;tag=as218b8918
    Call-ID: 49f8cf4a5a7fb68b6c88a3ae18b7298f@217.10.66.71
    CSeq: 102 INVITE
    User-Agent: Asterisk PBX
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
    Contact: <sip:9628879@68.227.169.158>
    Content-Type: application/sdp
    Content-Length: 209
    
    v=0
    o=root 2558 2558 IN IP4 68.227.169.158
    s=session
    c=IN IP4 68.227.169.158
    t=0 0
    m=audio 15630 RTP/AVP 3 0 8
    a=rtpmap:3 GSM/8000
    a=rtpmap:0 PCMU/8000
    a=rtpmap:8 PCMA/8000
    a=silenceSupp:off - - - -
    
    ---
        -- Attempting native bridge of SIP/217.10.66.71-09d10448 and SIP/3536-443b
    strubinsky*CLI>
    <-- SIP read from 217.116.119.252:5060:
    ACK sip:9628879@68.227.169.158 SIP/2.0
    Record-Route: <sip:9628879@217.116.119.252;ftag=as1b712083;lr=on>
    Max-Forwards:  9
    Via: SIP/2.0/UDP 217.116.119.252;branch=0
    Via: SIP/2.0/UDP 217.10.79.8;branch=0
    Via: SIP/2.0/UDP 217.10.66.71:5060;branch=z9hG4bK11696cd9
    From: "0414022926801" <sip:0414022926801@217.10.66.71>;tag=as1b712083
    To: <sip:4319628879@sipgate.net>;tag=as218b8918
    Contact: <sip:0414022926801@217.10.66.71>
    Call-ID: 49f8cf4a5a7fb68b6c88a3ae18b7298f@217.10.66.71
    CSeq: 102 ACK
    User-Agent: sipgate asterisk
    Content-Length: 0
    
    
    --- (13 headers 0 lines)---
    Destroying call '4113057f50ac696d2edd114f574254a9@127.0.0.1'
    Destroying call '66010d976234e4c72ffc4eb07534a508@127.0.0.1'
    Destroying call '2974ff062b7dd5684ec0512a27157685@127.0.0.1'
    strubinsky*CLI>
    <-- SIP read from 192.168.11.202:5060:
    BYE sip:0414022926801@192.168.11.69 SIP/2.0
    Via: SIP/2.0/UDP 192.168.11.202:5060;branch=z9hG4bK-cf15f509
    From: <sip:3536@192.168.11.202:5060>;tag=ba310c92db31bcc1i0
    To: "0414022926801" <sip:0414022926801@192.168.11.69>;tag=as51f3487a
    Call-ID: 1111d2c7243d4e4551fef97d65167ed9@192.168.11.69
    CSeq: 101 BYE
    Max-Forwards: 70
    User-Agent: Sipura/SPA2100-2.0.5(d)
    Content-Length: 0
    
    Vielen Dank für Deine Mühe!

    günter
     
  4. strubinsky

    strubinsky Neuer User

    Registriert seit:
    22 Feb. 2006
    Beiträge:
    3
    Zustimmungen:
    0
    Punkte für Erfolge:
    0
    Aaaahhhh!!!

    Ich habe die Situation analysiert und bin zu diesem Schluß gekommen:

    Wenn ich Diale ohne sound vorher zu haben, dann läutet das telefon und ich kann auf beiden Seiten sprechen und hören, das Freizeichen wird vom anrufenden apparat gehört

    Wenn Play, background, Musiconhold vor dem Dial kommen wird im trace zwar der jeweilige command abgespielt, aber es gibt keinen Ton (kein Freizeichen, absolut kein Ton). Kommt danach ein Dial, so läutet zwar das telefon aber wenn ich abhebe, dann kann weiterhin kein Ton gehört werden und der Dial Cmd timed out als ob ich gar nicht abgehoben hätte.

    Das sieht mir ganz so aus, als ob die beiden codecs nicht miteinander kompatibel sind.

    Sowohl mein Cisco 7960 als auch das Softphone funktionieren, was nicht so überraschend ist, da diese ja keinen Sound vor dem abheben machen. Das abheben scheint die richtigen codecs zu starten und zu synchronisieren.

    Kann wer mit einem sipgate.at account den Versuch machen:

    Code:
    exten => <Deine#>,1,Set(LANGUAGE()=de)
    exten => <Deine#>,n,Background(international-call)
    exten => <Deine#>,n,Dial(<Nebenstelle>)
    
    ... und rausfinden, ob "international call" zu hören ist. ( international-call.gsm ist ein standard soundfile in /asterisk-sounds-1.2.1/sounds/ =. Wenn sich das Verhaslten duplizieren läßt, dann haben die Österreicher was verbockt. Wenn nicht, dann muß ich weitersuchen (obwohl ich nicht mehr weiter weiß als in die C source zu gehen, was unglaublich aufwendig und zeitraubend ist; außer hier weiß jemand einen anderen Weg den ich versuchen kann bevor ich 'ans Eingemachte' gehe! )