Solved Call Forwarding with Snom results in 480

Discussion in '3CX Phone System - General' started by espritle, Feb 2, 2017.

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  1. espritle

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    Hi all,
    thanks for the great forum here!
    I'm right now struggling with CFU, CFB and CFNR with an Snom D725.
    When A calls B, B answers with 302 Moved Temporarily to the 3cx v15.
    But the 3cx sends out a 480 temporarily unavailable instead of an Invite in direction to C-Party.

    Status-Line: SIP/2.0 302 Moved Temporarily
    Message Header
    Via: SIP/2.0/UDP 192.168.0.8:5060;branch=z9hG4bK-524287-1---88c35149a0e97e42;rport=5060
    From: <sip:+49xxxxxxxx@192.168.0.8:5060;nf=e>;tag=79f55100
    To: <sip:32@192.168.0.8>;tag=yulzvj4jtc
    Call-ID: QimlVSk2Jn3u3rl5tqpToQ..
    CSeq: 1 INVITE
    User-Agent: snom725/8.7.5.8.11
    Contact: <sip:00xxxxxxxx@192.168.0.8;user=phone>
    Diversion: <sip:32@192.168.0.36:55016;line=0equtlqe>;reason="unconditional"
    Content-Length: 0

    Status-Line: SIP/2.0 480 Temporarily Unavailable
    Message Header
    Via: SIP/2.0/UDP 192.168.0.5;branch=z9hG4bK-3DED-75D
    Via: SIP/2.0/UDP 172.28.0.144:5060;branch=z9hG4bKg3Zqkv7ivmy8mqjiqxtlcfx7rsr7wre58
    To: "Anruf 0049xxxxxxx"<sip:+49xxxxxxxxxx@192.168.0.8:5060;user=phone>;tag=e9459a5d
    From: <sip:+49xxxxxxxxxx@192.168.0.5;user=phone>;tag=C37
    Call-ID: OA277F98F8498933984378552D9F0C
    CSeq: 1780 INVITE
    User-Agent: 3CXPhoneSystem 15.0.60903.0 (60903)
    Warning: 499 3cxlinux.xxxxxxxxxx.de "Terminated"
    Content-Length: 0

    Thanks in advance
    Klaus
     
  2. GiannosC_3CX

    GiannosC_3CX Guest

    Dear espritle,

    As far I see on your above logs (SIP/2.0 302 Moved Temporarily), you did a call forwarding from the snom725 device with IP address 192.168.0.36 and extension number 32 to Contact: <sip:00xxxxxxxx@192.168.0.8;user=phone>.

    This is a normal log. Also at the second log with 480 Temporarily Unavailable, indicate the problem is front of the 3CX Phone system.
    Please make sure that your gateway, voip provider, trunk configuration, max calls or your license etc.
     
  3. espritle

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    Dear Giannos,
    thanks a lot for Your reply.
    The 3cx is not even trying to send out the Invite, so the gateway and the voip provider is not yet involved in this scenario.
    I tried it out and it is possible to make two simultaneous calls. Also conference call with two outside phones is working well.

    In the trunk settings, I can't find any relevant setting, which could influence this behavior.
    For Reference, I copied the trunk settings below. (X means checked in checkbox, 0 means not checked)
    Trunk Details
    Enter name for Trunk
    Generic SIP Trunk
    Registrar/Server/Gateway Hostname or IP
    192.168.0.5
    5060
    Outbound Proxy
    192.168.0.5
    5060
    Number of SIM Calls
    10

    Authentication
    Type of Authentication
    Do not require - IP Based
    Authentication ID (aka SIP User ID)
    Authentication Password

    0 3 Way Authentication
    Routing of calls to Main Number
    Main Trunk No
    +49xxxxxxxxxxx
    Destination for calls during office hours
    33
    Destination for calls outside office hours
    33

    0 Set up Specific Office Hours for this trunk
    0 Play holiday prompt when it's a global holiday

    ===========================================================

    Call options
    X Allow inbound calls
    X Allow outbound calls
    X Disallow video calls

    Advanced
    X PBX Delivers Audio
    0 Supports Re-Invite
    0 Support Replaces
    0 Put Public IP in SIP VIA Header
    0 SRTP
    Re-Register Timeout
    0
    Select which IP to use in 'Contact' (SIP) and 'Connection'(SDP) fields
    Use Default Settings

    Codec Priority
    Add codecs Delete Move Up Move Down
    G.711 U-law
    G.711 A-law
    GSM-FR

    Thanks a lot
    Klaus
     
  4. GiannosC_3CX

    GiannosC_3CX Guest

    Hi Klaus,

    Please give us more information about your call flow. Also please update your Snom phone to supported min. firmware version.
     
  5. espritle

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    Hi Giannos,
    here are more details:
    192.168.0.5 is the gateway, 192.168.0.8 is the 3cx
    The snom phone works with different pbx in the same scenario ok. What is the minimum firmware version, 3cx needs for the D725?
    Do You need the 100 Trying and ACK as well?

    No. Time Source Destination Protocol Length Time to live Privacy Info
    1 0.000 192.168.0.5 192.168.0.8 SIP/SDP 1193 64 none Request: INVITE sip:+49bbbbbbb@192.168.0.8:5060;user=phone |
    2 0.153 192.168.0.8 192.168.0.5 SIP 415 64 Status: 100 Trying |
    3 0.204 192.168.0.8 192.168.0.36 SIP/SDP 991 64 Request: INVITE sip:32@192.168.0.36:55016;line=0equtlqe |
    4 0.224 192.168.0.36 192.168.0.8 SIP 415 64 Status: 100 Trying |
    5 0.244 192.168.0.36 192.168.0.8 SIP 494 64 Status: 302 Moved Temporarily |
    6 0.296 192.168.0.8 192.168.0.36 SIP 365 64 Request: ACK sip:32@192.168.0.36:55016;line=0equtlqe |
    7 0.328 192.168.0.8 192.168.0.5 SIP 559 64 Status: 480 Temporarily Unavailable |
    8 0.401 192.168.0.5 192.168.0.8 SIP 542 64 Request: ACK sip:+49bbbbbbbbb@192.168.0.8:5060;user=phone |



    Request-Line: INVITE sip:+498bbbbbbbb@192.168.0.8:5060;user=phone SIP/2.0
    Message Header
    Accept: application/dtmf-relay,application/media_control+xml,application/sdp,multipart/mixed
    Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
    Call-ID: OA277F98F8498933984378552D9F0C
    Contact: <sip:+498aaaaaaaa@192.168.0.5;transport=udp;user=phone>
    Content-Type: application/sdp
    CSeq: 1780 INVITE
    From: <sip:+498aaaaaaaaa@192.168.0.5;user=phone>;tag=C37
    Max-Forwards: 29
    Min-SE: 900
    P-Asserted-Identity: <sip:+498aaaaaaaaa@192.168.0.5;user=phone>
    Privacy: none
    Record-Route: <sip:192.168.0.5;lr>,<sip:172.30.129.21:5064;lr>
    Session-Expires: 1800;refresher=uac
    Supported: timer
    To: "Anruf 00498bbbbbbbb" <sip:+498bbbbbbb@192.168.0.8:5060;user=phone>
    Via: SIP/2.0/UDP 192.168.0.5;branch=z9hG4bK-3DED-75D,SIP/2.0/UDP 172.28.0.144:5060;branch=z9hG4bKg3Zqkv7ivmy8mqjiqxtlcfx7rsr7wre58
    Content-Length: 247
    Message Body
    Session Description Protocol
    Session Description Protocol Version (v): 0
    Owner/Creator, Session Id (o): BroadWorks 81239 1 IN IP4 192.168.0.5
    Session Name (s): -
    Connection Information (c): IN IP4 192.168.0.5
    Time Description, active time (t): 0 0
    Media Description, name and address (m): audio 11474 RTP/AVP 8 0 101
    Media Attribute (a): rtpmap:8 PCMA/8000
    Media Attribute (a): rtpmap:0 PCMU/8000
    Media Attribute (a): rtpmap:101 telephone-event/8000
    Media Attribute (a): fmtp:101 0-15
    Media Attribute (a): ptime:20
    Media Attribute (a): bsoft: 1 image udptl t
    ===========================================================================================

    Request-Line: INVITE sip:32@192.168.0.36:55016;line=0equtlqe SIP/2.0
    Message Header
    Via: SIP/2.0/UDP 192.168.0.8:5060;branch=z9hG4bK-524287-1---88c35149a0e97e42;rport
    Max-Forwards: 70
    Contact: <sip:+498aaaaaaa@192.168.0.8:5060>
    To: <sip:32@192.168.0.8>
    From: <sip:+498aaaaaaa@192.168.0.8:5060;nf=e>;tag=79f55100
    Call-ID: QimlVSk2Jn3u3rl5tqpToQ..
    CSeq: 1 INVITE
    Alert-Info: <http://www.notused.invalidtld>;info=external
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
    Content-Type: application/sdp
    Supported: replaces
    User-Agent: 3CXPhoneSystem 15.0.60903.0 (60903)
    Content-Length: 333
    Message Body
    Session Description Protocol
    Session Description Protocol Version (v): 0
    Owner/Creator, Session Id (o): 3cxPS 12956586217570304 29545769220440065 IN IP4 192.168.0.8
    Session Name (s): 3cxPS Audio call
    Connection Information (c): IN IP4 192.168.0.8
    Time Description, active time (t): 0 0
    Media Description, name and address (m): audio 7186 RTP/AVP 0 8 9 3 18 101
    Media Attribute (a): rtpmap:0 PCMU/8000
    Media Attribute (a): rtpmap:8 PCMA/8000
    Media Attribute (a): rtpmap:9 G722/8000
    Media Attribute (a): rtpmap:3 GSM/8000
    Media Attribute (a): rtpmap:18 G729/8000
    Media Attribute (a): fmtp:18 annexb=no
    Media Attribute (a): rtpmap:101 telephone-event/8000
    Media Attribute (a): sendre
    ===========================================================================================

    Status-Line: SIP/2.0 302 Moved Temporarily
    Message Header
    Via: SIP/2.0/UDP 192.168.0.8:5060;branch=z9hG4bK-524287-1---88c35149a0e97e42;rport=5060
    From: <sip:+498aaaaaaaa@192.168.0.8:5060;nf=e>;tag=79f55100
    To: <sip:32@192.168.0.8>;tag=yulzvj4jtc
    Call-ID: QimlVSk2Jn3u3rl5tqpToQ..
    CSeq: 1 INVITE
    User-Agent: snom725/8.7.5.8.11
    Contact: <sip:008ccccccc@192.168.0.8;user=phone>
    Diversion: <sip:32@192.168.0.36:55016;line=0equtlqe>;reason="unconditional"
    Content-Length: 0
    ===========================================================================================

    Status-Line: SIP/2.0 480 Temporarily Unavailable
    Message Header
    Via: SIP/2.0/UDP 192.168.0.5;branch=z9hG4bK-3DED-75D
    Via: SIP/2.0/UDP 172.28.0.144:5060;branch=z9hG4bKg3Zqkv7ivmy8mqjiqxtlcfx7rsr7wre58
    To: "Anruf 00498bbbbbbbb"<sip:+498bbbbbbbb@192.168.0.8:5060;user=phone>;tag=e9459a5d
    From: <sip:+498aaaaaaaa@192.168.0.5;user=phone>;tag=C37
    Call-ID: OA277F98F8498933984378552D9F0C
    CSeq: 1780 INVITE
    User-Agent: 3CXPhoneSystem 15.0.60903.0 (60903)
    Warning: 499 3cxlinux.xxxxxxxxxxx.de "Terminated"
    Content-Length: 0


    Thanks
    Klaus
     
  6. GiannosC_3CX

    GiannosC_3CX Guest

    Dear Klaus,

    I have sent you a p.m.
     
  7. GiannosC_3CX

    GiannosC_3CX Guest

    The issue was with outbound rules not matching and it is now solved.
     
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