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Thema: Sipcall.ch mit Asterisk und Snom

  1. #1
    Registriert seit
    19.07.2014
    Alter
    29
    Beiträge
    9

    Standard Sipcall.ch mit Asterisk und Snom

    Hallo Zusammen,
    Ich bin ganz neu auf dem Forum und hoffe ich habe alles richtig gemacht

    Mein Problem ist, ich kann zwar Angerufen werden jedoch kann ich niemanden anrufen und ich bin mit meinem latein am ende
    Meine
    Sip.conf:
    Code:
    [general]
    maxexpirey=1800
    defaultexpirey=120
    useragent=irgendwas
    externip=80.218.xxx.xxx
    disallow=all
    allow=ulaw
    encryption=no : turns on SRTP, if you have set this then the SIP device(s) MUST use it, it's either on or off, not optional
    
    register =>  4144xxxxxxxx:Passwort@pro2.voipgateway.org/4144xxxxxxx
    [4144xxxxxxx] ; sipcall
    defaultuser=4144xxxxxx
    ;callerid=4144xxxxxxxx <414xxxxxxx>
    type=peer
    remotesecret=1234
    qualify=yes
    ;insecure=very
    host=pro2.voipgateway.org
    fromuser=4144xxxxxxx
    fromdomain=pro2.voipgateway.org
    context=meine-telefone
    directmedia=no
    dtmfmode=info
    host=dynamic
    Meine Extensions.conf
    Code:
    [default]
    include => meine-telefone
    
    [meine-telefone]
    exten => 4144xxxx,1,Dial(SIP/4144xxxxxx)
    exten => _0.,1,Dial(SIP/${EXTEN}@4144xxxxxx,45,r)
    Wenn ich jemand anrufen möchte kommt dies:
    Code:
    <--- SIP read from UDP:192.168.61.21:2058 --->
    INVITE sip:0079xxxxxxx@192.168.61.91;user=phone SIP/2.0
    Via: SIP/2.0/UDP 192.168.61.21:2058;branch=z9hG4bK-z2lhb8abu458;rport
    From: "4144xxxxxxx" ;tag=29ttalmtl4
    To: 
    Call-ID: 53ca2fa91d35-uuust64opsqc
    CSeq: 1 INVITE
    Max-Forwards: 70
    Contact: ;reg-id=1
    X-Serialnumber: 000413250195
    P-Key-Flags: keys="3"
    User-Agent: snom300/8.7.3.25
    Accept: application/sdp
    Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE
    Allow-Events: talk, hold, refer, call-info
    Supported: timer, 100rel, replaces, from-change
    Session-Expires: 3600;refresher=uas
    Min-SE: 90
    Content-Type: application/sdp
    Content-Length: 190
    
    v=0
    o=root 1457399022 1457399022 IN IP4 192.168.61.21
    s=call
    c=IN IP4 192.168.61.21
    t=0 0
    m=audio 59474 RTP/AVP 0 8
    a=rtpmap:0 PCMU/8000
    a=rtpmap:8 PCMA/8000
    a=ptime:20
    a=sendrecv
    <------------->
    --- (19 headers 10 lines) ---
    Sending to 192.168.61.21:2058 (NAT)
    Using INVITE request as basis request - 53ca2fa91d35-uuust64opsqc
    Found peer '4144xxxxxxx' for '4144xxxxxxx' from 192.168.61.21:2058
    Found RTP audio format 0
    Found RTP audio format 8
    Found audio description format PCMU for ID 0
    Found audio description format PCMA for ID 8
    Capabilities: us - 0x4 (ulaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
    Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing)
    Peer audio RTP is at port 192.168.61.21:59474
    Looking for 0079xxxxxxx in meine-telefone (domain 192.168.61.91)
    list_route: hop: 
    
    <--- Transmitting (NAT) to 192.168.61.21:2058 --->
    SIP/2.0 100 Trying
    Via: SIP/2.0/UDP 192.168.61.21:2058;branch=z9hG4bK-z2lhb8abu458;received=192.168.61.21;rport=2058
    From: "4144xxxxxxx" ;tag=29ttalmtl4
    To: 
    Call-ID: 53ca2fa91d35-uuust64opsqc
    CSeq: 1 INVITE
    Server: irgendwas
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
    Supported: replaces, timer
    Session-Expires: 1800;refresher=uas
    Contact: 
    Content-Length: 0
    
    
    <------------>
    Audio is at 18594
    Adding codec 0x4 (ulaw) to SDP
    Reliably Transmitting (NAT) to 192.168.61.21:2058:
    INVITE sip:0079xxxxxxx@pro2.voipgateway.org SIP/2.0
    Via: SIP/2.0/UDP 192.168.61.91:5060;branch=z9hG4bK7f4139c7;rport
    Max-Forwards: 70
    From: "4144xxxxxxx" ;tag=as1d34ef28
    To: 
    Contact: 
    Call-ID: 0515a061392f5b2a1e2a94a20193fc9a@pro2.voipgateway.org
    CSeq: 102 INVITE
    User-Agent: irgendwas
    Date: Sat, 19 Jul 2014 08:43:21 GMT
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
    Supported: replaces, timer
    Content-Type: application/sdp
    Content-Length: 198
    
    v=0
    o=root 1808893510 1808893510 IN IP4 192.168.61.91
    s=Asterisk PBX 1.8.13.1~dfsg1-3+deb7u3
    c=IN IP4 192.168.61.91
    t=0 0
    m=audio 18594 RTP/AVP 0
    a=rtpmap:0 PCMU/8000
    a=ptime:20
    a=sendrecv
    
    ---
    
    <--- Transmitting (NAT) to 192.168.61.21:2058 --->
    SIP/2.0 180 Ringing
    Via: SIP/2.0/UDP 192.168.61.21:2058;branch=z9hG4bK-z2lhb8abu458;received=192.168.61.21;rport=2058
    From: "4144xxxxxxx" ;tag=29ttalmtl4
    To: ;tag=as7f5763e3
    Call-ID: 53ca2fa91d35-uuust64opsqc
    CSeq: 1 INVITE
    Server: irgendwas
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
    Supported: replaces, timer
    Session-Expires: 1800;refresher=uas
    Contact: 
    Content-Length: 0
    Wenn mir jemand helfen könnte währe ich sehr dankbar

  2. #2
    vstm ist offline Fortgeschrittener Benutzer
    Registriert seit
    28.12.2008
    Ort
    Züri Oberland
    Alter
    34
    Beiträge
    123

    Standard AW: Sipcall.ch mit Asterisk und Snom

    So wie dein Dialplan aussieht möchtest Du, dass man zuerst die 0 wählen muss um "raus zu wählen". Wenn Du dann "${EXTEN}" verwendest, ist dort die führende 0 ebenfalls drin, Du wählst also tatsächlich 0079xxxxxxx statt 079xxxxxxx. Um die führende 0 rauszuschneiden, kannst Du "${EXTEN:1}" verwenden.

    Kannst Du es mal so versuchen:

    Code:
    [meine-telefone]
    exten => 4144xxxx,1,Dial(SIP/4144xxxxxx)
    exten => _0.,1,Dial(SIP/${EXTEN:1}@4144xxxxxx,45,r)

  3. #3
    Registriert seit
    19.07.2014
    Alter
    29
    Beiträge
    9

    Standard AW: Sipcall.ch mit Asterisk und Snom

    Hallo,
    Vielen Dank für deine Antwort,
    Jetzt habe ich nun ein Anderes Problem,
    Dies wird nun bei set debug on angezeiget:
    (Wenn ich Raustelefonieren möchte)
    Code:
    <--- SIP read from UDP:192.168.61.21:2058 --->
    INVITE sip:00xxxxxxxxx@192.168.61.91;user=phone SIP/2.0
    Via: SIP/2.0/UDP 192.168.61.21:2058;branch=z9hG4bK-nzukczcumeus;rport
    From: "4144xxxxxxx" ;tag=dlfj53tk7d
    To: 
    Call-ID: 53cbeafbbeb0-2gmfp06nvu5q
    CSeq: 1 INVITE
    Max-Forwards: 70
    Contact: ;reg-id=1
    X-Serialnumber: 000413250195
    P-Key-Flags: keys="3"
    User-Agent: snom300/8.7.3.25
    Accept: application/sdp
    Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE
    Allow-Events: talk, hold, refer, call-info
    Supported: timer, 100rel, replaces, from-change
    Session-Expires: 3600;refresher=uas
    Min-SE: 90
    Content-Type: application/sdp
    Content-Length: 190
    
    v=0
    o=root 1514551418 1514551418 IN IP4 192.168.61.21
    s=call
    c=IN IP4 192.168.61.21
    t=0 0
    m=audio 54622 RTP/AVP 0 8
    a=rtpmap:0 PCMU/8000
    a=rtpmap:8 PCMA/8000
    a=ptime:20
    a=sendrecv
    <------------->
    --- (19 headers 10 lines) ---
    Sending to 192.168.61.21:2058 (NAT)
    Using INVITE request as basis request - 53cbeafbbeb0-2gmfp06nvu5q
    Found peer '4144xxxxxxx' for '4144xxxxxxx' from 192.168.61.21:2058
    Found RTP audio format 0
    Found RTP audio format 8
    Found audio description format PCMU for ID 0
    Found audio description format PCMA for ID 8
    Capabilities: us - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
    Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing)
    Peer audio RTP is at port 192.168.61.21:54622
    Looking for 00xxxxxxxxx in meine-telefone (domain 192.168.61.91)
    list_route: hop: 
    
    <--- Transmitting (NAT) to 192.168.61.21:2058 --->
    SIP/2.0 100 Trying
    Via: SIP/2.0/UDP 192.168.61.21:2058;branch=z9hG4bK-nzukczcumeus;received=192.168.61.21;rport=2058
    From: "4144xxxxxxx" ;tag=dlfj53tk7d
    To: 
    Call-ID: 53cbeafbbeb0-2gmfp06nvu5q
    CSeq: 1 INVITE
    Server: irgendwas
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
    Supported: replaces, timer
    Session-Expires: 1800;refresher=uas
    Contact: 
    Content-Length: 0
    
    
    <------------>
    Audio is at 14024
    Adding codec 0x4 (ulaw) to SDP
    Adding codec 0x2 (gsm) to SDP
    Adding codec 0x8 (alaw) to SDP
    Adding codec 0x800000000000 (testlaw) to SDP
    Reliably Transmitting (NAT) to 192.168.61.21:2058:
    INVITE sip:0xxxxxxxxx@pro2.voipgateway.org SIP/2.0
    Via: SIP/2.0/UDP 192.168.61.91:5060;branch=z9hG4bK46b00512;rport
    Max-Forwards: 70
    From: "4144xxxxxxx" ;tag=as2acfca9c
    To: 
    Contact: 
    Call-ID: 7515b2f935dad83c52cd7c051274a76c@pro2.voipgateway.org
    CSeq: 102 INVITE
    User-Agent: irgendwas
    Date: Sun, 20 Jul 2014 16:14:52 GMT
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
    Supported: replaces, timer
    Content-Type: application/sdp
    Content-Length: 245
    
    v=0
    o=root 1167233314 1167233314 IN IP4 192.168.61.91
    s=Asterisk PBX 1.8.13.1~dfsg1-3+deb7u3
    c=IN IP4 192.168.61.91
    t=0 0
    m=audio 14024 RTP/AVP 0 3 8
    a=rtpmap:0 PCMU/8000
    a=rtpmap:3 GSM/8000
    a=rtpmap:8 PCMA/8000
    a=ptime:20
    a=sendrecv
    
    ---
    
    <--- Transmitting (NAT) to 192.168.61.21:2058 --->
    SIP/2.0 180 Ringing
    Via: SIP/2.0/UDP 192.168.61.21:2058;branch=z9hG4bK-nzukczcumeus;received=192.168.61.21;rport=2058
    From: "4144xxxxxxx" ;tag=dlfj53tk7d
    To: ;tag=as49465220
    Call-ID: 53cbeafbbeb0-2gmfp06nvu5q
    CSeq: 1 INVITE
    Server: irgendwas
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
    Supported: replaces, timer
    Session-Expires: 1800;refresher=uas
    Contact: 
    Content-Length: 0
    
    
    <------------>
    Retransmitting #1 (NAT) to 192.168.61.21:2058:
    INVITE sip:0xxxxxxxxx@pro2.voipgateway.org SIP/2.0
    Via: SIP/2.0/UDP 192.168.61.91:5060;branch=z9hG4bK46b00512;rport
    Max-Forwards: 70
    From: "4144xxxxxxx" ;tag=as2acfca9c
    To: 
    Contact: 
    Call-ID: 7515b2f935dad83c52cd7c051274a76c@pro2.voipgateway.org
    CSeq: 102 INVITE
    User-Agent: irgendwas
    Date: Sun, 20 Jul 2014 16:14:52 GMT
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
    Supported: replaces, timer
    Content-Type: application/sdp
    Content-Length: 245
    
    v=0
    o=root 1167233314 1167233314 IN IP4 192.168.61.91
    s=Asterisk PBX 1.8.13.1~dfsg1-3+deb7u3
    c=IN IP4 192.168.61.91
    t=0 0
    m=audio 14024 RTP/AVP 0 3 8
    a=rtpmap:0 PCMU/8000
    a=rtpmap:3 GSM/8000
    a=rtpmap:8 PCMA/8000
    a=ptime:20
    a=sendrecv
    Da es zu viele Zeichen waren, habe ich den rest in einem txt file angehängt.
    Angehängte Dateien Angehängte Dateien

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