Problem with Ivcomming calls

Discussion in 'General' started by muertesbg, Jun 1, 2009.

  1. muertesbg

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    Hello,
    I have last version of elastix.
    The outgoing calls is OK, but I have a trouble with Incomming.
    The asterisk and X-lite soft phone is behind NAT.
    IP address of the Asterisk server is 10.0.0.240/255.255.255.0
    In my sip_general_custom.conf I add these lines:
    Code:
    externhost = XXX.XXX.XXX.XXX ; My External IP 
    externrefresh = 60
    bindport = 5060 ; Port to bind to (SIP is 5060)
    bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)
    localnet = 10.0.0.0/255.255.255.0
    
    Here is a snapshot of the Trunk and Inbounde Route:
    Picture 1
    Picture 2
    When I try to make a call from outside telephone, in my Asterisk*CLI nothing is comming.An other side is busy tone.
    Please help.
    After enable sip debug for peer:
    Code:
    <--- SIP read from 212.116.145.19:5060 --->
    INVITE sip:s@XXX.XXX.XXX.XXX SIP/2.0
    Record-Route: <sip:212.116.145.19;lr=on;ftag=5B282A04-3E0;nat=yes>
    Via: SIP/2.0/UDP 212.116.145.19;branch=z9hG4bKedf6.0c327f7.0
    Via: SIP/2.0/UDP 212.116.145.6:5060;rport=52681;branch=z9hG4bKBDD881534
    Remote-Party-ID: "35932XXXXXX " <sip:35932XXXXXX@ITDNET>;party=calling;screen=no;privacy=off
    From: "359XXXXXXX " <sip:35932XXXXX@ITDNET>;tag=5B282A04-3E0
    To: <sip:35932XXXXXX@212.116.145.19>
    Date: Mon, 01 Jun 2009 13:27:30 GMT
    Call-ID: CAB30EFA-4DE611DE-B486EB50-9800209F@ITDNET
    Supported: 100rel,timer,resource-priority,replaces
    Min-SE:  1800
    Cisco-Guid: 3400172442-1306923486-3028413264-2550145183
    User-Agent: Cisco-SIPGateway/IOS-12.x
    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
    CSeq: 101 INVITE
    Timestamp: 1243862850
    Contact: <sip:35932XXXXXXX@212.116.145.6:5060>
    Expires: 180
    Allow-Events: telephone-event
    Max-Forwards: 67
    Content-Type: application/sdp
    Content-Disposition: session;handling=required
    Content-Length: 330
    
    v=0
    o=CiscoSystemsSIP-GW-UserAgent 7744 2716 IN IP4 212.116.145.6
    s=SIP Call
    c=IN IP4 212.116.145.6
    t=0 0
    m=audio 18316 RTP/AVP 8 0 18 101
    c=IN IP4 212.116.145.6
    a=rtpmap:8 PCMA/8000
    a=rtpmap:0 PCMU/8000
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=no
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    a=direction:passive
    
    <------------->
    --- (23 headers 14 lines) ---
    Sending to 212.116.145.19 : 5060 (no NAT)
    Using INVITE request as basis request - CAB30EFA-4DE611DE-B486EB50-9800209F@ITDNET
    Found peer 'CoolBox511'
    Found RTP audio format 8
    Found RTP audio format 0
    Found RTP audio format 18
    Found RTP audio format 101
    Peer audio RTP is at port 212.116.145.6:18316
    Found audio description format PCMA for ID 8
    Found audio description format PCMU for ID 0
    Found audio description format G729 for ID 18
    Found audio description format telephone-event for ID 101
    Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
    Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
    Peer audio RTP is at port 212.116.145.6:18316
    Looking for s in from-CoolBox511 (domain XXX.XXX.XXX.XXX (My IP))
    
    <--- Reliably Transmitting (no NAT) to 212.116.145.19:5060 --->
    SIP/2.0 404 Not Found
    Via: SIP/2.0/UDP 212.116.145.19;branch=z9hG4bKedf6.0c327f7.0;received=212.116.145.19
    Via: SIP/2.0/UDP 212.116.145.6:5060;rport=52681;branch=z9hG4bKBDD881534
    From: "35932XXXXXX " <sip:35932XXXXXX@ITDNET>;tag=5B282A04-3E0
    To: <sip:35932XXXXXX@212.116.145.19>;tag=as35ad7e9c
    Call-ID: CAB30EFA-4DE611DE-B486EB50-9800209F@ITDNET
    CSeq: 101 INVITE
    User-Agent: Asterisk PBX
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
    Supported: replaces
    Content-Length: 0
    
    
    <------------>
    Scheduling destruction of SIP dialog 'CAB30EFA-4DE611DE-B486EB50-9800209F@ITDNET' in 32000 ms (Method: INVITE)
    ubuntu*CLI>
    <--- SIP read from 212.116.145.19:5060 --->
    ACK sip:s@XXX.XXX.XXX.XXX (My IP) SIP/2.0
    Via: SIP/2.0/UDP 212.116.145.19;branch=z9hG4bKedf6.0c327f7.0
    From: "35932XXXXXX " <sip:35932XXXXXX@ITDNET>;tag=5B282A04-3E0
    Call-ID: CAB30EFA-4DE611DE-B486EB50-9800209F@ITDNET
    To: <sip:35932XXXXXX@212.116.145.19>;tag=as35ad7e9c
    CSeq: 101 ACK
    Max-Forwards: 70
    Content-Length: 0
    
    
    <------------->
    --- (8 headers 0 lines) ---
    Really destroying SIP dialog 'CAB30EFA-4DE611DE-B486EB50-9800209F@ITDNET' Method: ACK
    
    In my log file is:
    Code:
    [Jun  1 16:26:18] NOTICE[5310] chan_sip.c: Call from '35932XXXXXX' to extension 's' rejected because extension not found.
    
     
  2. muertesbg

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    The Problem is SOLVED.
     
  3. ramoncio

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    Could you explain how you did it?
    Maybe it can help some others..
     
  4. muertesbg

    Joined:
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    Yes of course.My mistake.
    In the Peer Details change context=from-trunk, also in the User Details again context=from trunk.
    This is solve my problems.Now Incoming and Outgoig calls working OK.
    Thank You very much for Your help.
     

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