[Gelöst] WLAN Telefone keine Funktion mehr nach Update auf 4.5

  • Hallo Community,


    wir haben ein großes Problem: Seit dem automatischen Update auf Vers. 4.5 streiken unsere WLAN-Telefone ( Siemens SL75) komplett. Man kann wählen, doch sobald das Gespräch angenommen wird, wird die Verbindung automatisch beendet. Danach hängen sich die Geräte zumeist auf.
    Festnetz und DECT-Telefone funktionieren einwandfrei.
    Logs und Shell liefern keine aufschlussreichen Informationen.
    Hat jemand eine Idee oder ähnliche Probleme?


    Gruß


    Niklas

  • Hallo slu,
    hier ein Anruf :


    -- Executing [621@international:1] AGI("SIP/HellwigMobil-000001b2", "agi://localhost/initdial.agi") in new stack
    -- AGI Script Executing Application: (noop) Options: (---[info]---new Phone: SIP/HellwigMobil )
    -- AGI Script Executing Application: (Set) Options: (__SFCLIDNUM=1411631)
    -- AGI Script Executing Application: (Set) Options: (__SFCLIDINTNUM=631)
    -- AGI Script Executing Application: (Set) Options: (__SFCLIDNAME=Niklas Hellwig)
    -- AGI Script Executing Application: (noop) Options: (---[info]---Call from : (name) Niklas Hellwig (num) 1411631 (intern) 631 )
    -- AGI Script Executing Application: (Set) Options: (CHANNEL(language)=de)
    -- AGI Script Executing Application: (SetMusicOnHold) Options: (IDSP)
    Extension Changed a1030[hint] new state Busy for Notify User Strobel
    Extension Changed a1030[hint] new state Busy for Notify User Meyer
    Extension Changed a1030[hint] new state Busy for Notify User Balkhi
    Extension Changed a1030[hint] new state Busy for Notify User HeyneHome (queued)
    Extension Changed a1030[hint] new state Busy for Notify User Zentrale
    Extension Changed a1030[hint] new state Busy for Notify User Wagner
    Extension Changed a1030[hint] new state Busy for Notify User AID.K.Meyer
    Extension Changed a1030[hint] new state Busy for Notify User Heyne
    Extension Changed a1030[hint] new state Busy for Notify User Zabel
    Extension Changed a1030[hint] new state Busy for Notify User Rothmann
    Extension Changed a1030[hint] new state Busy for Notify User Lange
    Extension Changed a1030[hint] new state Busy for Notify User Lauenstein
    Extension Changed a1030[hint] new state Busy for Notify User Buettner
    -- AGI Script Executing Application: (Set) Options: (CDR(userfield)=434025)
    -- AGI Script Executing Application: (setCallerPres) Options: (ALLOWED)
    -- AGI Script Executing Application: (Set) Options: (CALLERID(all)=621: Niklas Hellwig 631 <631>)
    -- AGI Script Executing Application: (noop) Options: (---[AGIKernel]---PostTargetDeterminationIncoming:Calling module hook 'Zentrale DATAIDENT')
    -- AGI Script Executing Application: (noop) Options: (---[AGIKernel]---PostTargetDeterminationIncoming:Calling module hook 'IDSP Zentrale')
    -- AGI Script Executing Application: (noop) Options: (---[AGIKernel]---PostTargetDeterminationIncoming:Calling module hook 'Hotline' )
    -- AGI Script Executing Application: (noop) Options: (---[Info]---Call incoming to 621 )
    -- AGI Script Executing Application: (Set) Options: (CDR(userfield)=434025)
    -- AGI Script Executing Application: (setCallerPres) Options: (ALLOWED)
    -- AGI Script Executing Application: (Set) Options: (CALLERID(all)=621: Niklas Hellwig 631 <631>)
    -- AGI Script Executing Application: (noop) Options: (---[AGIKernel]---PostPhoneSelection:Calling module hook 'Gruppenanzeige' )
    -- AGI Script Executing Application: (ChanIsAvail) Options: (SIP/Strobel.Mobil)
    -- AGI Script Executing Application: (ChanIsAvail) Options: (SIP/Strobel)
    -- Hungup 'Zap/20-1'
    == Spawn extension (international, 001714748588, 1) exited non-zero on 'SIP/Lange-000001b1'
    -- Channel 0/2, span 7 received AOC-E charging 7 units
    -- AGI Script Executing Application: (SIPAddHeader) Options: (X-STARFACE-TTL: 10)
    -- AGI Script Executing Application: (SIPAddHeader) Options: (X-STARFACE-CallerAccountId: 1030)
    -- AGI Script Executing Application: (SIPAddHeader) Options: (X-STARFACE-SiteUUID: 4f47692f-50ea-4219-8649-3ac7c6562cbc)
    -- AGI Script Executing Application: (Set) Options: (CDR(accountcode)=1)
    -- AGI Script Executing Application: (Dial) Options: (SIP/Strobel.Mobil&SIP/Strobel|45|wW)
    -- Called Strobel.Mobil
    -- Called Strobel
    -- SIP/Strobel-000001b6 is ringing
    Extension Changed a1241[hint] new state Idle for Notify User Heyne
    Extension Changed a1241[hint] new state Idle for Notify User Balkhi
    Extension Changed a1032[hint] new state Ringing for Notify User HeyneHome (queued)
    Extension Changed a1032[hint] new state Ringing for Notify User AID.K.Meyer
    Extension Changed a1032[hint] new state Ringing for Notify User Meyer
    Extension Changed a1032[hint] new state Ringing for Notify User Rothmann
    Extension Changed a1032[hint] new state Ringing for Notify User Bauer
    Extension Changed a1032[hint] new state Ringing for Notify User Zentrale
    Extension Changed a1032[hint] new state Ringing for Notify User Weber
    Extension Changed a1032[hint] new state Ringing for Notify User Hellwig
    Extension Changed a1032[hint] new state Ringing for Notify User Schulz
    Extension Changed a1032[hint] new state Ringing for Notify User Heyne
    Extension Changed a1032[hint] new state Ringing for Notify User Buettner
    Extension Changed a1032[hint] new state Ringing for Notify User Wagner
    -- SIP/Strobel-000001b6 is ringing
    -- SIP/Strobel-000001b6 is ringing
    -- SIP/Strobel-000001b6 is ringing
    -- Remote UNIX connection
    -- Remote UNIX connection disconnected
    -- SIP/Strobel-000001b6 answered SIP/HellwigMobil-000001b2

  • Von "SIP/HellwigMobil" (SL75) auf" SIP/Stobel".


    Klingelt auch ganz normal, nur in dem Moment wo abgehoben wird bricht die Verbindung ab.

  • Ich war auf dem falschen Weg, es ist nicht das selber Problem wie mit den PAP2.
    Was passiert denn wenn Du einen externen Teilnehmer anrufst?
    Gibts ein Firmwareupdate für die SL75?

  • Das Problem besteht sobald eines der SL75 involviert ist intern/extern, Festnetz/DECT anrufen/annehmen egal.
    Die SL75 haben alle die neuste Firmware drauf. :(

  • Wie geht denn der Trace weiter? Nach dem


    Zitat

    SIP/Strobel-000001b6 answered SIP/HellwigMobil-000001b2


    kommt bestimmt noch etwas interessantes...

    Gruß / Regards
    Philipp

  • Wollen wirs hoffen ;)
    hier die nächsten paar sekunden:


    -- Remote UNIX connection disconnected
    -- SIP/Strobel-000001b6 answered SIP/HellwigMobil-000001b2
    Extension Changed a1032[hint] new state Busy for Notify User HeyneHome (queued)
    Extension Changed a1032[hint] new state Busy for Notify User AID.K.Meyer
    Extension Changed a1032[hint] new state Busy for Notify User Meyer
    Extension Changed a1032[hint] new state Busy for Notify User Rothmann
    Extension Changed a1032[hint] new state Busy for Notify User Bauer
    Extension Changed a1032[hint] new state Busy for Notify User Zentrale
    Extension Changed a1032[hint] new state Busy for Notify User Weber
    Extension Changed a1032[hint] new state Busy for Notify User Hellwig
    Extension Changed a1032[hint] new state Busy for Notify User Schulz
    Extension Changed a1032[hint] new state Busy for Notify User Heyne
    Extension Changed a1032[hint] new state Busy for Notify User Buettner
    Extension Changed a1032[hint] new state Busy for Notify User Wagner
    -- Got SIP response 405 "Method Not Allowed" back from 10.1.6.156
    == Spawn extension (international, 621, 1) exited non-zero on 'SIP/HellwigMobil-000001b2'
    Extension Changed a1032[hint] new state Idle for Notify User HeyneHome (queued)
    Extension Changed a1032[hint] new state Idle for Notify User AID.K.Meyer
    Extension Changed a1032[hint] new state Idle for Notify User Meyer
    Extension Changed a1032[hint] new state Idle for Notify User Rothmann
    Extension Changed a1032[hint] new state Idle for Notify User Bauer
    Extension Changed a1032[hint] new state Idle for Notify User Zentrale
    Extension Changed a1032[hint] new state Idle for Notify User Weber
    Extension Changed a1032[hint] new state Idle for Notify User Hellwig
    Extension Changed a1032[hint] new state Idle for Notify User Schulz
    Extension Changed a1032[hint] new state Idle for Notify User Heyne
    Extension Changed a1032[hint] new state Idle for Notify User Buettner
    Extension Changed a1032[hint] new state Idle for Notify User Wagner
    Extension Changed a1030[hint] new state Idle for Notify User Strobel
    Extension Changed a1030[hint] new state Idle for Notify User Meyer
    Extension Changed a1030[hint] new state Idle for Notify User Balkhi
    Extension Changed a1030[hint] new state Idle for Notify User HeyneHome (queued)
    Extension Changed a1030[hint] new state Idle for Notify User Zentrale
    Extension Changed a1030[hint] new state Idle for Notify User Wagner
    Extension Changed a1030[hint] new state Idle for Notify User AID.K.Meyer
    Extension Changed a1030[hint] new state Idle for Notify User Heyne
    Extension Changed a1030[hint] new state Idle for Notify User Zabel
    Extension Changed a1030[hint] new state Idle for Notify User Rothmann
    Extension Changed a1030[hint] new state Idle for Notify User Lange
    Extension Changed a1030[hint] new state Idle for Notify User Lauenstein
    Extension Changed a1030[hint] new state Idle for Notify User Buettner
    -- Accepting overlap voice call from '04052865418' to '1411' on channel 0/1, span 6
    -- Starting simple switch on 'Zap/16-1'
    [Aug 30 11:58:27] DTMF[11516]: channel.c:2382 __ast_read: DTMF end '2' received on Zap/16-1, duration 0 ms
    [Aug 30 11:58:27] DTMF[11516]: channel.c:2424 __ast_read: DTMF end accepted without begin '2' on Zap/16-1
    [Aug 30 11:58:27] DTMF[11516]: channel.c:2435 __ast_read: DTMF end passthrough '2' on Zap/16-1
    [Aug 30 11:58:27] DTMF[11516]: channel.c:2382 __ast_read: DTMF end '6' received on Zap/16-1, duration 0 ms
    [Aug 30 11:58:27] DTMF[11516]: channel.c:2424 __ast_read: DTMF end accepted without begin '6' on Zap/16-1
    [Aug 30 11:58:27] DTMF[11516]: channel.c:2435 __ast_read: DTMF end passthrough '6' on Zap/16-1
    [Aug 30 11:58:28] DTMF[11516]: channel.c:2382 __ast_read: DTMF end '1' received on Zap/16-1, duration 0 ms
    [Aug 30 11:58:28] DTMF[11516]: channel.c:2424 __ast_read: DTMF end accepted without begin '1' on Zap/16-1
    [Aug 30 11:58:28] DTMF[11516]: channel.c:2435 __ast_read: DTMF end passthrough '1' on Zap/16-1
    -- Executing [1411261@dataident-incoming:1] Set("Zap/16-1", "channelname=dataident-incoming") in new stack
    -- Executing [1411261@dataident-incoming:2] Goto("Zap/16-1", "incoming|1411261|1") in new stack
    -- Goto (incoming,1411261,1)
    -- Executing [1411261@incoming:1] Goto("Zap/16-1", "calling|1411261|1") in new stack
    -- Goto (calling,1411261,1)
    -- Executing [1411261@calling:1] AGI("Zap/16-1", "agi://localhost/initdial.agi") in new stack
    -- AGI Script Executing Application: (noop) Options: (---[info]---new Phone: ZAP/16-1 )
    -- AGI Script Executing Application: (Set) Options: (__SFCLIDNUM=+494052865418)
    -- AGI Script Executing Application: (Set) Options: (__SFCLIDINTNUM=)
    -- AGI Script Executing Application: (Set) Options: (__SFCLIDNAME=Andrew Köppke, CASIO)
    -- AGI Script Executing Application: (noop) Options: (---[info]---Call from : (name) Andrew Köppke, CASIO (num) +494052865418 (intern) )
    -- AGI Script Executing Application: (Set) Options: (CHANNEL(language)=de)
    -- AGI Script Executing Application: (set) Options: (addedtrunk=true)
    -- AGI Script Executing Application: (noop) Options: (---[debug]---Adding Trunk to callerid 000494052865418 )
    -- AGI Script Executing Application: (Set) Options: (CDR(userfield)=434026)
    -- AGI Script Executing Application: (setCallerPres) Options: (ALLOWED)
    -- AGI Script Executing Application: (Set) Options: (CALLERID(all)=1411261: Andrew Köppke, CASIO 000494052865418 <000494052865418>)
    -- AGI Script Executing Application: (noop) Options: (---[AGIKernel]---PostTargetDeterminationIncoming:Calling module hook 'Zentrale DATAIDENT')
    -- AGI Script Executing Application: (noop) Options: (---[AGIKernel]---PostTargetDeterminationIncoming:Calling module hook 'IDSP Zentrale')
    -- AGI Script Executing Application: (noop) Options: (---[AGIKernel]---PostTargetDeterminationIncoming:Calling module hook 'Hotline' )
    -- AGI Script Executing Application: (noop) Options: (---[Info]---Call incoming to 1411261 )
    -- AGI Script Executing Application: (Set) Options: (CDR(userfield)=434026)
    -- AGI Script Executing Application: (setCallerPres) Options: (ALLOWED)
    -- AGI Script Executing Application: (Set) Options: (CALLERID(all)=1411261: Andrew Köppke, CASIO 000494052865418 <000494052865418>)
    -- AGI Script Executing Application: (noop) Options: (---[AGIKernel]---PostPhoneSelection:Calling module hook 'Gruppenanzeige' )
    -- AGI Script Executing Application: (ChanIsAvail) Options: (SIP/Roeschmann)
    -- Accepting overlap voice call from '04121296178' to '1411' on channel 0/1, span 4
    -- Starting simple switch on 'Zap/10-1'
    [Aug 30 11:58:32] DTMF[11517]: channel.c:2382 __ast_read: DTMF end '2' received on Zap/10-1, duration 0 ms
    [Aug 30 11:58:32] DTMF[11517]: channel.c:2424 __ast_read: DTMF end accepted without begin '2' on Zap/10-1
    [Aug 30 11:58:32] DTMF[11517]: channel.c:2435 __ast_read: DTMF end passthrough '2' on Zap/10-1
    -- AGI Script Executing Application: (ChanIsAvail) Options: (SIP/Roeschmann.Mobil)
    -- AGI Script Executing Application: (Set) Options: (CDR(accountcode)=1)
    -- AGI Script Executing Application: (Dial) Options: (SIP/Roeschmann|15|wW)

  • ähem, ja!
    Stimmt, das ist wohl interessant, oder?


    habs überprüft, kommt bei allen Telefonaten:


    Got SIP response 405 "Method Not Allowed" back from "IP-Adresse des Anrufers"


    bzw:


    AGI Script Executing Application: (noop) Options: (---[Debug]---Hangupcause: NOTDEFINED Dialstatus: ANSWER )
    -- AGI Script agi://localhost/initdial.agi completed, returning 0
    -- Executing [621@international:2] Hangup("SIP/Hellwig-00000033", "127") in new stack
    == Spawn extension (international, 621, 2) exited non-zero on 'SIP/Hellwig-00000033'


    wenn ich von Festnetz auf Mobil anrufe

  • geb mal auf der asterisk CLI folgendes ein


    Zitat

    sip debug peer HellwigMobil


    - führe erneut einen arnuf aus
    - poste das Log


    Deaktivieren geht mit "sip no debug"

    Gruß / Regards
    Philipp

  • Hier das Log, aufgrund der Länge in zwei Posts und leicht gekürzt


    dataident*CLI> sip debug peer HellwigMobil
    SIP Debugging Enabled for IP: 10.1.6.156:5060


    <--- SIP read from 10.1.6.156:5060 --->
    INVITE sip:601@10.1.1.20 SIP/2.0
    Max-Forwards: 70
    Content-Length: 285
    Via: SIP/2.0/UDP 10.1.6.156:5060;branch=z9hG4bK0a52344ce
    Call-ID: 20e050abacc189f
    From: HellwigMobil <sip:HellwigMobil@10.1.1.20>;tag=119b674c5a46dd4
    To: sip:601@10.1.1.20
    CSeq: 300601947 INVITE
    Supported: timer
    Session-Expires: 7200
    Allow-Events: talk, hold, conference
    Allow: INVITE,ACK,CANCEL,BYE,REFER,NOTIFY,MESSAGE,INFO
    Content-Type: application/sdp
    Contact: HellwigMobil <sip:HellwigMobil@10.1.6.156:5060;transport=udp>
    Supported: replaces
    User-Agent: Gigaset SL75 WLAN M5T SIP-UA SAFE/v3.6.4.8


    v=0
    o=MxSIP 0 477433675 IN IP4 10.1.6.156
    s=SIP Call
    c=IN IP4 10.1.6.156
    t=0 0
    m=audio 10000 RTP/AVP 8 0 4 18 9 101
    a=rtpmap:8 PCMA/8000
    a=rtpmap:0 PCMU/8000
    a=rtpmap:4 G723/8000
    a=rtpmap:18 G729/8000
    a=rtpmap:9 G722/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15


    <------------->
    --- (16 headers 13 lines) ---
    Sending to 10.1.6.156 : 5060 (NAT)
    Using INVITE request as basis request - 20e050abacc189f


    <--- Reliably Transmitting (no NAT) to 10.1.6.156:5060 --->
    SIP/2.0 407 Proxy Authentication Required
    Via: SIP/2.0/UDP 10.1.6.156:5060;branch=z9hG4bK0a52344ce;received=10.1.6.156
    From: HellwigMobil <sip:HellwigMobil@10.1.1.20>;tag=119b674c5a46dd4
    To: sip:601@10.1.1.20;tag=as40487651
    Call-ID: 20e050abacc189f
    CSeq: 300601947 INVITE
    User-Agent: STARFACE PBX
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
    Supported: replaces
    Proxy-Authenticate: Digest algorithm=MD5, realm="STARFACE", nonce="668cc7b2"
    Content-Length: 0



    <------------>
    Scheduling destruction of SIP dialog '20e050abacc189f' in 32000 ms (Method: INVITE)
    Found user 'HellwigMobil'


    <--- SIP read from 10.1.6.156:5060 --->
    ACK sip:601@10.1.1.20 SIP/2.0
    Max-Forwards: 70
    Content-Length: 0
    Via: SIP/2.0/UDP 10.1.6.156:5060;branch=z9hG4bK0a52344ce
    Call-ID: 20e050abacc189f
    From: HellwigMobil <sip:HellwigMobil@10.1.1.20>;tag=119b674c5a46dd4
    To: sip:601@10.1.1.20;tag=as40487651
    CSeq: 300601947 ACK
    User-Agent: Gigaset SL75 WLAN M5T SIP-UA SAFE/v3.6.4.8



    <------------->
    --- (9 headers 0 lines) ---


    <--- SIP read from 10.1.6.156:5060 --->
    INVITE sip:601@10.1.1.20 SIP/2.0
    Max-Forwards: 70
    Content-Length: 285
    Via: SIP/2.0/UDP 10.1.6.156:5060;branch=z9hG4bK4c489b544
    Call-ID: 20e050abacc189f
    From: HellwigMobil <sip:HellwigMobil@10.1.1.20>;tag=119b674c5a46dd4
    To: sip:601@10.1.1.20
    CSeq: 300601948 INVITE
    Supported: timer
    Session-Expires: 7200
    Allow-Events: talk, hold, conference
    Allow: INVITE,ACK,CANCEL,BYE,REFER,NOTIFY,MESSAGE,INFO
    Content-Type: application/sdp
    Proxy-Authorization:Digest response="eb8e9403881f05c67e534364886eabe7",username="HellwigMobil",realm="STARFACE",nonce="668cc7b2",algorithm=MD5,uri="sip:601@10.1.1.20"
    Supported: replaces
    Contact: HellwigMobil <sip:HellwigMobil@10.1.6.156:5060;transport=udp>
    User-Agent: Gigaset SL75 WLAN M5T SIP-UA SAFE/v3.6.4.8


    v=0
    o=MxSIP 0 477433675 IN IP4 10.1.6.156
    s=SIP Call
    c=IN IP4 10.1.6.156
    t=0 0
    m=audio 10000 RTP/AVP 8 0 4 18 9 101
    a=rtpmap:8 PCMA/8000
    a=rtpmap:0 PCMU/8000
    a=rtpmap:4 G723/8000
    a=rtpmap:18 G729/8000
    a=rtpmap:9 G722/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15


    <------------->
    --- (17 headers 13 lines) ---
    Sending to 10.1.6.156 : 5060 (no NAT)
    Using INVITE request as basis request - 20e050abacc189f
    Found user 'HellwigMobil'
    Found RTP audio format 8
    Found RTP audio format 0
    Found RTP audio format 4
    Found RTP audio format 18
    Found RTP audio format 9
    Found RTP audio format 101
    Found audio description format PCMA for ID 8
    Found audio description format PCMU for ID 0
    Found audio description format G723 for ID 4
    Found audio description format G729 for ID 18
    Found audio description format G722 for ID 9
    Found audio description format telephone-event for ID 101
    Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x110d (g723|ulaw|alaw|g729|g722)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
    Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x1 (telephone-event), combined - 0x0 (nothing)
    Peer audio RTP is at port 10.1.6.156:10000
    Looking for 601 in international (domain 10.1.1.20)
    list_route: hop: <sip:HellwigMobil@10.1.6.156:5060;transport=udp>


    <--- Transmitting (no NAT) to 10.1.6.156:5060 --->
    SIP/2.0 100 Trying
    Via: SIP/2.0/UDP 10.1.6.156:5060;branch=z9hG4bK4c489b544;received=10.1.6.156
    From: HellwigMobil <sip:HellwigMobil@10.1.1.20>;tag=119b674c5a46dd4
    To: sip:601@10.1.1.20
    Call-ID: 20e050abacc189f
    CSeq: 300601948 INVITE
    User-Agent: STARFACE PBX
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
    Supported: replaces
    Contact: <sip:601@10.1.1.20>
    Content-Length: 0



    <------------>
    -- Executing [601@international:1] AGI("SIP/HellwigMobil-00000093", "agi://localhost/initdial.agi") in new stack
    -- AGI Script Executing Application: (noop) Options: (---[info]---new Phone: SIP/HellwigMobil )
    -- AGI Script Executing Application: (Set) Options: (__SFCLIDNUM=1411631)
    -- AGI Script Executing Application: (Set) Options: (__SFCLIDINTNUM=631)
    -- AGI Script Executing Application: (Set) Options: (__SFCLIDNAME=Niklas Hellwig)
    -- AGI Script Executing Application: (noop) Options: (---[info]---Call from : (name) Niklas Hellwig (num) 1411631 (intern) 631 )
    -- AGI Script Executing Application: (Set) Options: (CHANNEL(language)=de)

    -- AGI Script Executing Application: (SetMusicOnHold) Options: (IDSP)
    -- AGI Script Executing Application: (Set) Options: (CDR(userfield)=434112)
    -- AGI Script Executing Application: (setCallerPres) Options: (ALLOWED)
    -- AGI Script Executing Application: (Set) Options: (CALLERID(all)=601: Niklas Hellwig 631 <631>)
    -- AGI Script Executing Application: (noop) Options: (---[AGIKernel]---PostTargetDeterminationIncoming:Calling module hook 'Zentrale DATAIDENT')
    -- AGI Script Executing Application: (noop) Options: (---[AGIKernel]---PostTargetDeterminationIncoming:Calling module hook 'IDSP Zentrale')
    -- AGI Script Executing Application: (noop) Options: (---[AGIKernel]---PostTargetDeterminationIncoming:Calling module hook 'Hotline' )
    -- AGI Script Executing Application: (noop) Options: (---[Info]---Call incoming to 601 )
    -- AGI Script Executing Application: (Set) Options: (CDR(userfield)=434112)
    -- AGI Script Executing Application: (setCallerPres) Options: (ALLOWED)
    -- AGI Script Executing Application: (Set) Options: (CALLERID(all)=601: Niklas Hellwig 631 <631>)
    -- AGI Script Executing Application: (noop) Options: (---[AGIKernel]---PostPhoneSelection:Calling module hook 'Gruppenanzeige' )
    -- AGI Script Executing Application: (ChanIsAvail) Options: (SIP/Meyer)
    -- AGI Script Executing Application: (ChanIsAvail) Options: (SIP/Memobil2)
    -- AGI Script Executing Application: (ChanIsAvail) Options: (SIP/Me.Mobil)
    -- AGI Script Executing Application: (SIPAddHeader) Options: (X-STARFACE-TTL: 10)
    -- AGI Script Executing Application: (SIPAddHeader) Options: (X-STARFACE-CallerAccountId: 1030)
    -- AGI Script Executing Application: (SIPAddHeader) Options: (X-STARFACE-SiteUUID: 4f47692f-50ea-4219-8649-3ac7c6562cbc)
    -- AGI Script Executing Application: (Set) Options: (CDR(accountcode)=1)
    -- AGI Script Executing Application: (Dial) Options: (SIP/Meyer&SIP/Me.Mobil|45|wW)
    -- Called Meyer
    -- Called Me.Mobil
    -- SIP/Meyer-00000096 is ringing

  • Teil Zwei:


    <--- Transmitting (no NAT) to 10.1.6.156:5060 --->
    SIP/2.0 180 Ringing
    Via: SIP/2.0/UDP 10.1.6.156:5060;branch=z9hG4bK4c489b544;received=10.1.6.156
    From: HellwigMobil <sip:HellwigMobil@10.1.1.20>;tag=119b674c5a46dd4
    To: sip:601@10.1.1.20;tag=as205f0917
    Call-ID: 20e050abacc189f
    CSeq: 300601948 INVITE
    User-Agent: STARFACE PBX
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
    Supported: replaces
    Contact: <sip:601@10.1.1.20>
    Content-Length: 0




    -- SIP/Me.Mobil-00000097 is ringing
    -- SIP/Meyer-00000096 is ringing
    -- SIP/Meyer-00000096 is ringing
    -- SIP/Meyer-00000096 is ringing


    <--- SIP read from 10.1.6.156:5060 --->


    <------------->
    -- SIP/Meyer-00000096 answered SIP/HellwigMobil-00000093
    Audio is at 10.1.1.20 port 19998
    Adding codec 0x8 (alaw) to SDP
    Adding codec 0x4 (ulaw) to SDP


    <--- Reliably Transmitting (no NAT) to 10.1.6.156:5060 --->
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 10.1.6.156:5060;branch=z9hG4bK4c489b544;received=10.1.6.156
    From: HellwigMobil <sip:HellwigMobil@10.1.1.20>;tag=119b674c5a46dd4
    To: sip:601@10.1.1.20;tag=as205f0917
    Call-ID: 20e050abacc189f
    CSeq: 300601948 INVITE
    User-Agent: STARFACE PBX
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
    Supported: replaces
    Contact: <sip:601@10.1.1.20>
    Content-Type: application/sdp
    Content-Length: 175


    v=0
    o=root 15971 15971 IN IP4 10.1.1.20
    s=session
    c=IN IP4 10.1.1.20
    t=0 0
    m=audio 19998 RTP/AVP 8 0
    a=rtpmap:8 PCMA/8000
    a=rtpmap:0 PCMU/8000
    a=ptime:20
    a=sendrecv



    set_destination: Parsing <sip:HellwigMobil@10.1.6.156:5060;transport=udp> for address/port to send to
    set_destination: set destination to 10.1.6.156, port 5060
    Audio is at 10.1.1.20 port 19998
    Adding codec 0x8 (alaw) to SDP
    Adding codec 0x4 (ulaw) to SDP
    14 headers, 10 lines
    Reliably Transmitting (no NAT) to 10.1.6.156:5060:
    UPDATE sip:HellwigMobil@10.1.6.156:5060;transport=udp SIP/2.0
    Via: SIP/2.0/UDP 10.1.1.20:5060;branch=z9hG4bK32cda3b1;rport
    From: sip:601@10.1.1.20;tag=as205f0917
    To: HellwigMobil <sip:HellwigMobil@10.1.1.20>;tag=119b674c5a46dd4
    Contact: <sip:601@10.1.1.20>
    Call-ID: 20e050abacc189f
    CSeq: 102 UPDATE
    User-Agent: STARFACE PBX
    Max-Forwards: 70
    P-Asserted-Identity: "Henning Meyer 601" <sip:601@10.1.1.20>
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
    X-asterisk-info: SIP-callerid-update
    Content-Type: application/sdp
    Content-Length: 175


    v=0
    o=root 15971 15972 IN IP4 10.1.1.20
    s=session
    c=IN IP4 10.1.1.20
    t=0 0
    m=audio 19998 RTP/AVP 8 0
    a=rtpmap:8 PCMA/8000
    a=rtpmap:0 PCMU/8000
    a=ptime:20
    a=sendrecv


    ---


    <--- SIP read from 10.1.6.156:5060 --->
    ACK sip:601@10.1.1.20 SIP/2.0
    Max-Forwards: 70
    Content-Length: 0
    Via: SIP/2.0/UDP 10.1.6.156:5060;branch=z9hG4bKa4567b9b7
    Call-ID: 20e050abacc189f
    From: HellwigMobil <sip:HellwigMobil@10.1.1.20>;tag=119b674c5a46dd4
    To: sip:601@10.1.1.20;tag=as205f0917
    CSeq: 300601948 ACK
    Proxy-Authorization:Digest response="eb8e9403881f05c67e534364886eabe7",username="HellwigMobil",realm="STARFACE",nonce="668cc7b2",algorithm=MD5,uri="sip:601@10.1.1.20"
    User-Agent: Gigaset SL75 WLAN M5T SIP-UA SAFE/v3.6.4.8



    <------------->
    --- (10 headers 0 lines) ---


    <--- SIP read from 10.1.6.156:5060 --->
    SIP/2.0 405 Method Not Allowed
    Call-ID: 20e050abacc189f
    CSeq: 102 UPDATE
    From: sip:601@10.1.1.20;tag=as205f0917
    To: HellwigMobil <sip:HellwigMobil@10.1.1.20>;tag=119b674c5a46dd4
    Via: SIP/2.0/UDP 10.1.1.20:5060;branch=z9hG4bK32cda3b1;rport
    Content-Length: 0
    Allow: INVITE,ACK,CANCEL,BYE,REFER,NOTIFY,MESSAGE,INFO
    Allow: INVITE
    Allow: ACK
    Allow: CANCEL
    Allow: BYE
    Allow: REFER
    Allow: NOTIFY
    Allow: MESSAGE
    Allow: INFO
    User-Agent: Gigaset SL75 WLAN M5T SIP-UA SAFE/v3.6.4.8



    <------------->
    --- (17 headers 0 lines) ---
    -- Got SIP response 405 "Method Not Allowed" back from 10.1.6.156


    <--- SIP read from 10.1.6.156:5060 --->
    BYE sip:601@10.1.1.20 SIP/2.0
    Max-Forwards: 70
    Content-Length: 0
    Via: SIP/2.0/UDP 10.1.6.156:5060;branch=z9hG4bK7a4688d36
    Call-ID: 20e050abacc189f
    From: HellwigMobil <sip:HellwigMobil@10.1.1.20>;tag=119b674c5a46dd4
    To: sip:601@10.1.1.20;tag=as205f0917
    CSeq: 300601949 BYE
    Supported: timer
    Proxy-Authorization:Digest response="dea207fad50d5a7240a7f1dd902d8932",username="HellwigMobil",realm="STARFACE",nonce="668cc7b2",algorithm=MD5,uri="sip:601@10.1.1.20"
    Supported: replaces
    User-Agent: Gigaset SL75 WLAN M5T SIP-UA SAFE/v3.6.4.8



    <------------->
    --- (12 headers 0 lines) ---
    Sending to 10.1.6.156 : 5060 (no NAT)


    <--- Transmitting (no NAT) to 10.1.6.156:5060 --->
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 10.1.6.156:5060;branch=z9hG4bK7a4688d36;received=10.1.6.156
    From: HellwigMobil <sip:HellwigMobil@10.1.1.20>;tag=119b674c5a46dd4
    To: sip:601@10.1.1.20;tag=as205f0917
    Call-ID: 20e050abacc189f
    CSeq: 300601949 BYE
    User-Agent: STARFACE PBX
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
    Supported: replaces
    Content-Length: 0



    <------------>
    == Spawn extension (international, 601, 1) exited non-zero on 'SIP/HellwigMobil-00000093'


    Really destroying SIP dialog '20e050abacc189f' Method: BYE
    -- Accepting overlap voice call from '04075249960' to '1411281' on channel 0/1, span 4
    -- Starting simple switch on 'Zap/10-1'

  • Danke für den Trace. Das Telefon antwortet auf das von der STARFACE versendete UPDATE mit eben dieser Fehlermeldung "405 Method Not Allowed"


    Die Verwendung von UPDATES ist in der 4.5 hinzugekommen, um z.B. die CallerID dynamisch abändern zu können. Dies wird u.a. bei Umleitungen benötigt.
    Eine Deaktivierung dieses Mechanismus ist ad-hoc nicht möglich, ich nehme diesen Vorgang aber bei uns im System auf.

    Gruß / Regards
    Philipp

  • Wir können, mit anderen Worten, derzeit also nichts tun, z.B Updates ausschalten?
    Tritt das Problem nur bei den SL75 auf, hat kein anderer das Problem?
    Die "Method not allowed"-Antwort haben die Telefone früher aber auch schon geschickt, wenn ich mich recht erinnere!?
    Besteht vielleicht die Möglichkeit eines Downgrades bis dieses Problem behoben wird?
    Einspielen eines Backups von vor 4.5 bringts leider nicht, da bleibt die FW-Version bestehen.

  • Hallo Philipp,


    gibt es eine Möglichkeit die WLAN Telefone wieder nutzbar zu machen? Derzeit sind alle mobilen Telefone nicht nutzbar und das ist verständlicherweise sehr problematisch bei 30 Usern :(


    Gruß Henning

  • Hallo Nickhell, ids,


    seit Version 4.5 gibt es das Feature "Aktualisierung der Rufnummer des Anrufers am Endgerät nach erfolgter Rufvermittlung". Dieses Realisieren wir mit SIP-Nachrichten im Moment an dem das Gespräch entgegen genommen wird. Diese entsprechen zwar dem SIP-Protokoll, können aber scheinbar nicht von allen Telefonen sauber verarbeitet werden. Diese Geräte antworten auf den Empfang dieser Nachrichten anstatt dem erwarteten "200 OK" mit einen "405 Method Not Allowed". Diesen Error-Code vom Telefon interpretiert der Asterisk wiederum als einen schwerwiegenden Fehler und quittiert den Vorgang mit einem Gesprächsabbruch.


    Dieses fehlerträchtige Verhalten haben wir mit den von uns empfohlenen (und getesteten) Endgeräten bislang nicht beobachten können. Wir untersuchen dieses Problem augenblicklich mich Nachdruck und planen, bis Mitte nächster Woche ein Update für die 4.5 bereit zu stellen, das Kunden mit solchen Endgeräten weiter hilft. Bis dahin können wir leider keinen Workaround anbieten.

    mfg
    may

    Einmal editiert, zuletzt von may ()

  • May's Ausführungen sind leider so nicht ganz korrekt.


    Im hier genannten Fall schickt das ENDGERÄT die BYE Nachricht, nicht die STARFACE. Bis zur Response 405 des Endgerätes ist soweit alles RFC-konform. Warum danach ein Rufabbau durch das Endgerät erfolgt ist mir völlig schleierhaft und muss einmal mit Siemens geklärt werden.


    Welche Softwareversion kommt in den SL75 Geräten zum Einsatz? Sind diese von Siemens noch unterstützt?


    UPDATE: Eine Recherche zeigt dass dies keine Siemens, sondern Gigaset Endgeräte sind. Bitte zuallererst nun sicherstellen, dass auf den Geräten tatsächlich die letzte verfügbare Firmware aufgespielt ist:


    http://gigaset.com/de/de/cms/P…cesDownloadsSoftware.html (Drop-Down Gigaset SL75 WLAN)


    Dies am Besten mit zwei ausgewählten Endgeräten testen.


    Die Geräte sind wohl recht alt und vom Hersteller nicht mehr aktiv vertrieben oder unterstützt.


    Viele Grüsse,
    Martin

    Einmal editiert, zuletzt von mkoenig ()

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