Incoming calls not working Elastix 1.6.2-2

Discussion in 'General' started by eiger3970, May 28, 2010.

  1. eiger3970

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    Hi Elastix forum,
    after an upgrade to Elastix 1.6.2-2, my incoming calls aren't working.
    All trunks and routes are the same as on my old Elastix version.

    I have repeated a yum update but still not working. My asterisk -r command displays the following from a mobile incoming call to the PBX:
    <------------>
    -- Executing [xxxxxxxxxx@from-sip-external:1] NoOp("SIP/xxxxxxxxxx-08a97668", "Received incoming SIP connection from unknown peer to xxxxxxxxxx") in new stack
    -- Executing [xxxxxxxxxx@from-sip-external:2] Set("SIP/xxxxxxxxxx-08a97668", "DID=xxxxxxxxxx") in new stack
    -- Executing [xxxxxxxxxx@from-sip-external:3] Goto("SIP/xxxxxxxxxx-08a97668", "s|1") in new stack
    -- Goto (from-sip-external,s,1)
    -- Executing [s@from-sip-external:1] GotoIf("SIP/xxxxxxxxxx-08a97668", "0?from-trunk|0755938793|1") in new stack
    -- Executing [s@from-sip-external:2] Set("SIP/xxxxxxxxxx-08a97668", "TIMEOUT(absolute)=15") in new stack
    -- Channel will hangup at 2010-05-29 03:04:21 UTC.
    -- Executing [s@from-sip-external:3] Answer("SIP/xxxxxxxxxx-08a97668", "") in new stack
    Audio is at 192.168.1.103 port 18718
    Adding codec 0x8 (alaw) to SDP
    Adding codec 0x4 (ulaw) to SDP
    Adding codec 0x400 (ilbc) to SDP
    Adding non-codec 0x1 (telephone-event) to SDP

    <--- Reliably Transmitting (NAT) to 202.169.178.10:5060 --->
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 202.169.178.10:5060;branch=z9hG4bK1caa9b20a4c008425-d099-0;received=202.169.178.10
    From: "xxxxxxxxxx"<sip:xxxxxxxxxx@202.169.178.10:5060>;tag=700f73f8-co534-INS001
    To: <sip:xxxxxxxxxx@192.168.1.103:5060>;tag=as79267859
    Call-ID: 1f05-409-724197052737-TCP-IMG03-3-210.80.190.202
    CSeq: 53401 INVITE
    User-Agent: Asterisk PBX
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
    Supported: replaces
    Contact: <sip:xxxxxxxxxx@192.168.1.103>
    Content-Type: application/sdp
    Content-Length: 309

    v=0
    o=root 2744 2744 IN IP4 192.168.1.103
    s=session
    c=IN IP4 192.168.1.103
    t=0 0
    m=audio 18718 RTP/AVP 8 0 96 101
    a=rtpmap:8 PCMA/8000
    a=rtpmap:0 PCMU/8000
    a=rtpmap:96 iLBC/8000
    a=fmtp:96 mode=30
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=silenceSupp:eek:ff - - - -
    a=ptime:20
    a=sendrecv

    <------------>
    -- Executing [s@from-sip-external:4] Wait("SIP/xxxxxxxxxx-08a97668", "2") in new stack
    xxxxxxxxxx*CLI>
    <--- SIP read from 202.169.178.10:5060 --->
    ACK sip:xxxxxxxxxx@121.208.10.172:5060 SIP/2.0
    Via: SIP/2.0/UDP 202.169.178.10:5060;branch=z9hG4bK1caa9b20a4c008425-d099-1
    Max-Forwards: 70
    To: <sip:xxxxxxxxxx@192.168.1.103:5060>;tag=as79267859
    From: "xxxxxxxxxx"<sip:xxxxxxxxxx@202.169.178.10:5060>;tag=700f73f8-co534-INS001
    Call-ID: 1f05-409-724197052737-TCP-IMG03-3-210.80.190.202
    CSeq: 53401 ACK
    User-Agent: ENSR2.5.4
    Content-Length: 0


    <------------->
    --- (9 headers 0 lines) ---
    -- Executing [s@from-sip-external:5] Playback("SIP/xxxxxxxxxx-08a97668", "ss-noservice") in new stack
    -- <SIP/xxxxxxxxxx-08a97668> Playing 'ss-noservice' (language 'en')
    -- Executing [s@from-sip-external:6] PlayTones("SIP/xxxxxxxxxx-08a97668", "congestion") in new stack
    == Spawn extension (from-sip-external, s, 6) exited non-zero on 'SIP/xxxxxxxxxx-08a97668'
    -- Executing [h@from-sip-external:1] NoOp("SIP/xxxxxxxxxx-08a97668", "Hangup") in new stack
    -- Executing [h@from-sip-external:2] Set("SIP/xxxxxxxxxx-08a97668", "DID=s") in new stack
    -- Executing [h@from-sip-external:3] Goto("SIP/xxxxxxxxxx-08a97668", "s|1") in new stack
    -- Goto (from-sip-external,s,1)
    -- Executing [s@from-sip-external:1] GotoIf("SIP/xxxxxxxxxx-08a97668", "0?from-trunk|s|1") in new stack
    -- Executing [s@from-sip-external:2] Set("SIP/xxxxxxxxxx-08a97668", "TIMEOUT(absolute)=15") in new stack
    -- Channel will hangup at 2010-05-29 03:04:28 UTC.
    -- Executing [s@from-sip-external:3] Answer("SIP/xxxxxxxxxx-08a97668", "") in new stack
    == Spawn extension (from-sip-external, s, 3) exited non-zero on 'SIP/xxxxxxxxxx-08a97668'
    Scheduling destruction of SIP dialog '1f05-409-724197052737-TCP-IMG03-3-210.80.190.202' in 6400 ms (Method: ACK)
    set_destination: Parsing <sip:xxxxxxxxxx@202.169.178.10:5060> for address/port to send to
    set_destination: set destination to 202.169.178.10, port 5060
    Reliably Transmitting (NAT) to 202.169.178.10:5060:
    BYE sip:xxxxxxxxxx@202.169.178.10:5060 SIP/2.0
    Via: SIP/2.0/UDP 192.168.1.103:5060;branch=z9hG4bK1bc42afd;rport
    From: <sip:xxxxxxxxxx@192.168.1.103:5060>;tag=as79267859
    To: "xxxxxxxxxx"<sip:0403420075@202.169.178.10:5060>;tag=700f73f8-co534-INS001
    Call-ID: 1f05-409-724197052737-TCP-IMG03-3-210.80.190.202
    CSeq: 102 BYE
    User-Agent: Asterisk PBX
    Max-Forwards: 70
    X-Asterisk-HangupCause: Unknown
    X-Asterisk-HangupCauseCode: 0
    Content-Length: 0
     
  2. danardf

    Joined:
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    Hi.

    If you do a simple submit on every incoming routes?
    Or maybe to remake them?
     
  3. eiger3970

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    Thanks DanardF, after the reinstallation, I found I had missed the Elastix\PBX\General Settings\Allow Anonymous Inbound SIP Calls? settings which was on no. Changed to yes and all fixed!

    The only final issue is the poor quality of the VoIP calls as the reinstallation removed my G.729 codec. I really don't want to pay another $10 for the G.729 codec.

    The ilbc codec is free but it's only source code and I need object code. Does Elastix have a download for ilbc as object code?
     
  4. Patrick_elx

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    you probably want to create a proper inbound route for all your trunk and avoid treat the anonymous call as an exception if you really need it or completely disable it.

    If you had a license for your g729 codecs previously, you can reinstall them with the same license key you had before.


    ilbc can not be provided compiled anymore to not infringe the license.
    But if you do a seach on this forum, you will find a compiled version of it that you can just copy to the proper folder.
     
  5. eiger3970

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    Well, finally fixed the the g729 by finding another post showing how to download the binaries. Thanks to the help which ruled out a few steps to take me to the fix.
     

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