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Dropped Calls

Discussion in '3CX Phone System - General' started by coreybrett, Jun 3, 2010.

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  1. coreybrett

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    My phone system is randomly dropping calls.

    I have 3CXv8 installed on an XP VM in Hyper-V.

    I'm using Callcentric as my VOIP provider.

    I don't see any errors in the logs (3CX or phones) or the event viewer on the server.

    I've had this happen to me as well as others, audio stops, then the call disconnects.

    Screen shots and log files can be found here...
    http://dl.dropbox.com/u/3594910/3cx_files.zip

    You can see in the 3CX log that call 83 was disconnected.
     
  2. leejor

    leejor Well-Known Member

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    You probably need to use something like Wireshark to determine which end is terminating the call. Can your provider shed any light on it?
     
  3. coreybrett

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    On further inspection, it seems that the phone itself is terminating the call.
    In the log below from the phones SIP trace, the second to last paragraph is the phone sending a BYE message.
    I'm just not sure why it's doing this.
    I have Snom 320 phones with v8.2.29 FW.

    Can anyone help me?

    Code:
    Sent to udp:192.168.76.19:5060 at 4/6/2010 14:29:05:379 (1099 bytes):
    
    INVITE sip:5405743345@192.168.76.19:5060;user=phone SIP/2.0
    Via: SIP/2.0/UDP 192.168.76.217:2057;branch=z9hG4bK-ckv75erq72rc;rport
    From: "Karen Fox" <sip:217@192.168.76.19:5060>;tag=e2dqdx93fa
    To: <sip:5405743345@192.168.76.19:5060;user=phone>
    Call-ID: 3c30e4d7100e-ahlkqj8poqiu
    CSeq: 1 INVITE
    Max-Forwards: 70
    Contact: <sip:217@192.168.76.217:2057;line=ktegtk7d>;reg-id=1
    X-Serialnumber: 000413271B23
    P-Key-Flags: keys="3"
    User-Agent: snom320/8.2.29
    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: 318
    
    v=0
    o=root 1800499861 1800499861 IN IP4 192.168.76.217
    s=call
    c=IN IP4 192.168.76.217
    t=0 0
    m=audio 63300 RTP/AVP 0 8 3 18 101
    a=rtpmap:0 pcmu/8000
    a=rtpmap:8 pcma/8000
    a=rtpmap:3 gsm/8000
    a=rtpmap:18 g729/8000
    a=fmtp:18 annexb=no
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=ptime:20
    a=sendrecv
    
    Received from udp:192.168.76.19:5060 at 4/6/2010 14:29:05:538 (485 bytes):
    
    SIP/2.0 407 Proxy Authentication Required
    Via: SIP/2.0/UDP 192.168.76.217:2057;branch=z9hG4bK-ckv75erq72rc;rport=2057
    Proxy-Authenticate: Digest nonce="414d535c0219d6e221:371c91930b6d97f342d670dab3e79951",algorithm=MD5,realm="3CXPhoneSystem"
    To: <sip:5405743345@192.168.76.19:5060;user=phone>;tag=2e7c172a
    From: "Karen Fox"<sip:217@192.168.76.19:5060>;tag=e2dqdx93fa
    Call-ID: 3c30e4d7100e-ahlkqj8poqiu
    CSeq: 1 INVITE
    User-Agent: 3CXPhoneSystem 8.0.10708.0
    Content-Length: 0
    
    Sent to udp:192.168.76.19:5060 at 4/6/2010 14:29:05:551 (409 bytes):
    
    ACK sip:5405743345@192.168.76.19:5060;user=phone SIP/2.0
    Via: SIP/2.0/UDP 192.168.76.217:2057;branch=z9hG4bK-ckv75erq72rc;rport
    From: "Karen Fox" <sip:217@192.168.76.19:5060>;tag=e2dqdx93fa
    To: <sip:5405743345@192.168.76.19:5060;user=phone>;tag=2e7c172a
    Call-ID: 3c30e4d7100e-ahlkqj8poqiu
    CSeq: 1 ACK
    Max-Forwards: 70
    Contact: <sip:217@192.168.76.217:2057;line=ktegtk7d>;reg-id=1
    Content-Length: 0
    
    Sent to udp:192.168.76.19:5060 at 4/6/2010 14:29:05:570 (1335 bytes):
    
    INVITE sip:5405743345@192.168.76.19:5060;user=phone SIP/2.0
    Via: SIP/2.0/UDP 192.168.76.217:2057;branch=z9hG4bK-iwe7ko9e6ofo;rport
    From: "Karen Fox" <sip:217@192.168.76.19:5060>;tag=e2dqdx93fa
    To: <sip:5405743345@192.168.76.19:5060;user=phone>
    Call-ID: 3c30e4d7100e-ahlkqj8poqiu
    CSeq: 2 INVITE
    Max-Forwards: 70
    Contact: <sip:217@192.168.76.217:2057;line=ktegtk7d>;reg-id=1
    X-Serialnumber: 000413271B23
    P-Key-Flags: keys="3"
    User-Agent: snom320/8.2.29
    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
    Proxy-Authorization: Digest username="217",realm="3CXPhoneSystem",nonce="414d535c0219d6e221:371c91930b6d97f342d670dab3e79951",uri="sip:5405743345@192.168.76.19:5060;user=phone",response="9f111ed907b2468c95c802a1378b29d3",algorithm=MD5
    Content-Type: application/sdp
    Content-Length: 318
    
    v=0
    o=root 1800499861 1800499861 IN IP4 192.168.76.217
    s=call
    c=IN IP4 192.168.76.217
    t=0 0
    m=audio 63300 RTP/AVP 0 8 3 18 101
    a=rtpmap:0 pcmu/8000
    a=rtpmap:8 pcma/8000
    a=rtpmap:3 gsm/8000
    a=rtpmap:18 g729/8000
    a=fmtp:18 annexb=no
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=ptime:20
    a=sendrecv
    
    Received from udp:192.168.76.19:5060 at 4/6/2010 14:29:05:707 (285 bytes):
    
    SIP/2.0 100 Trying
    Via: SIP/2.0/UDP 192.168.76.217:2057;branch=z9hG4bK-iwe7ko9e6ofo;rport=2057
    To: <sip:5405743345@192.168.76.19:5060;user=phone>
    From: "Karen Fox" <sip:217@192.168.76.19:5060>;tag=e2dqdx93fa
    Call-ID: 3c30e4d7100e-ahlkqj8poqiu
    CSeq: 2 INVITE
    Content-Length: 0
    
    Received from udp:192.168.76.19:5060 at 4/6/2010 14:29:07:561 (772 bytes):
    
    SIP/2.0 180 Ringing
    Via: SIP/2.0/UDP 192.168.76.217:2057;branch=z9hG4bK-iwe7ko9e6ofo;rport=2057
    Contact: <sip:5405743345@192.168.76.19:5060;user=phone>
    To: <sip:5405743345@192.168.76.19:5060;user=phone>;tag=5b07d513
    From: "Karen Fox"<sip:217@192.168.76.19:5060>;tag=e2dqdx93fa
    Call-ID: 3c30e4d7100e-ahlkqj8poqiu
    CSeq: 2 INVITE
    Content-Type: application/sdp
    User-Agent: 3CXPhoneSystem 8.0.10708.0
    Content-Length: 344
    
    v=0
    o=3cxPS 530747228160 425453420545 IN IP4 192.168.76.19
    s=3cxPS Audio call
    c=IN IP4 192.168.76.19
    t=0 0
    m=audio 7028 RTP/AVP 0 8 3 18 101
    c=IN IP4 192.168.76.19
    a=rtpmap:0 PCMU/8000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:3 GSM/8000
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=no
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=sendrecv
    
    Received from udp:192.168.76.19:5060 at 4/6/2010 14:29:20:174 (849 bytes):
    
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 192.168.76.217:2057;branch=z9hG4bK-iwe7ko9e6ofo;rport=2057
    Contact: <sip:5405743345@192.168.76.19:5060>
    To: <sip:5405743345@192.168.76.19:5060;user=phone>;tag=5b07d513
    From: "Karen Fox"<sip:217@192.168.76.19:5060>;tag=e2dqdx93fa
    Call-ID: 3c30e4d7100e-ahlkqj8poqiu
    CSeq: 2 INVITE
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
    Content-Type: application/sdp
    User-Agent: 3CXPhoneSystem 8.0.10708.0
    Content-Length: 344
    
    v=0
    o=3cxPS 530747228160 425453420545 IN IP4 192.168.76.19
    s=3cxPS Audio call
    c=IN IP4 192.168.76.19
    t=0 0
    m=audio 7028 RTP/AVP 0 8 3 18 101
    c=IN IP4 192.168.76.19
    a=rtpmap:0 PCMU/8000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:3 GSM/8000
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=no
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=sendrecv
    
    Sent to udp:192.168.76.19:5060 at 4/6/2010 14:29:20:206 (398 bytes):
    
    ACK sip:5405743345@192.168.76.19:5060 SIP/2.0
    Via: SIP/2.0/UDP 192.168.76.217:2057;branch=z9hG4bK-hc5pvvlzrs4p;rport
    From: "Karen Fox" <sip:217@192.168.76.19:5060>;tag=e2dqdx93fa
    To: <sip:5405743345@192.168.76.19:5060;user=phone>;tag=5b07d513
    Call-ID: 3c30e4d7100e-ahlkqj8poqiu
    CSeq: 2 ACK
    Max-Forwards: 70
    Contact: <sip:217@192.168.76.217:2057;line=ktegtk7d>;reg-id=1
    Content-Length: 0
    
    Sent to udp:192.168.76.19:5060 at 4/6/2010 14:29:44:976 (570 bytes):
    
    BYE sip:5405743345@192.168.76.19:5060 SIP/2.0
    Via: SIP/2.0/UDP 192.168.76.217:2057;branch=z9hG4bK-jug5ltv3ptut;rport
    From: "Karen Fox" <sip:217@192.168.76.19:5060>;tag=e2dqdx93fa
    To: <sip:5405743345@192.168.76.19:5060;user=phone>;tag=5b07d513
    Call-ID: 3c30e4d7100e-ahlkqj8poqiu
    CSeq: 3 BYE
    Max-Forwards: 70
    Contact: <sip:217@192.168.76.217:2057;line=ktegtk7d>;reg-id=1
    User-Agent: snom320/8.2.29
    RTP-RxStat: Total_Rx_Pkts=1357,Rx_Pkts=1357,Rx_Pkts_Lost=0,Remote_Rx_Pkts_Lost=0
    RTP-TxStat: Total_Tx_Pkts=1939,Tx_Pkts=1939,Remote_Tx_Pkts=0
    Content-Length: 0
    
    Received from udp:192.168.76.19:5060 at 4/6/2010 14:29:45:193 (376 bytes):
    
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 192.168.76.217:2057;branch=z9hG4bK-jug5ltv3ptut;rport=2057
    Contact: <sip:5405743345@192.168.76.19:5060>
    To: <sip:5405743345@192.168.76.19:5060;user=phone>;tag=5b07d513
    From: "Karen Fox"<sip:217@192.168.76.19:5060>;tag=e2dqdx93fa
    Call-ID: 3c30e4d7100e-ahlkqj8poqiu
    CSeq: 3 BYE
    User-Agent: 3CXPhoneSystem 8.0.10708.0
    Content-Length: 0
    
     
  4. leejor

    leejor Well-Known Member

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    This is one solution that came up....http://forum.snom.com/index.php?showtopic=2197&s=b463ba060701f18b1e9c8091808e1583
     
  5. coreybrett

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    Ok, so I don't think it's the phones after all.

    I was one the phone with an outside caller and the audio (both ways) stopped, but the call itself didn't terminate. Someone else in another office was on another call as well, and their call audio failed at the same time. So, whatever it is, seems to effect all calls at the same time.

    I have the firewall on the 3CX box shutoff.

    Again, I can't find any errors in the event log or in the 3CX log.
     
  6. leejor

    leejor Well-Known Member

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    If other calls are being affected at the same time then it could be a device, or load on your network. It could be your ISP.
     
  7. coreybrett

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    Well, I had 3CX running in a XP VM on Hyper-V, now I have it running on an old Celeron with a gig of memory. No more issues... running great!
     
  8. CaffeinatedTech

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    Hmm, I am having the same issue as you described, and am running 3CX v8 in a XP VM on Hyper-V Server 2008 R2. Nothing obvious in the logs, and support is reluctant to assist as I'm using an "unsupported" gateway (SPA3000) and VoIP provider.

    I've had both inbound and outbound calls go silent then drop (probably other end hanging up) on both PSTN and VoIP calls.

    I've given the VM maximum priority, 2 cores, but could only spare 512MB of RAM for it. There isn't anything network intensive running on any of the other VMs and the CPU isn't maxing out. I don't get any other network related issues.

    I might see if I can scrounge up an old box to run it on and see if that solves the problem. Pity though, its much easier to manage a VM than a physical box.
     
  9. coreybrett

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    Well I'm kinda glad that I'm not the only one to experience this.
    I don't have any empirical evidence that the VM was the issue, but the success rate of calls has sky-rocketed since moving to a real box. Last Friday we had at least 6 disconnected calls, and this week so far only 2.

    I do believe I have some power issues in my server room, so I haven't ruled that out as a possible issue yet. Now I'm kinda wondering if the Hyper-V host was more sensitive to power fluctuations somehow. Or maybe the switch was part of the problem.

    Since I couldn't find a single error in any of the logs (Hyper-V host, guest VM or phone), troubleshooting has been a bit of a guessing game.
     
  10. mfm

    mfm Active Member

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    Hi,

    It doesnt have to be a guessing game, if you want to diagnose an issue that is of this complexting you have to breakdown into litte pieces and find out where the problem piece is.

    First grab a wireshark capture and see who is dropping the call. If it is the gateway/ voip provider then you must look into the device/ provider as to why they are dropping the call, psobbily contact them, If it is the PBX then you need to start sifting trough errors in the more in depth 3cxphone trace logs, these logs give in depth information of issues, with that in hand and an indication of where the problem is we can look further into it,
     
    Stop hovering to collapse... Click to collapse... Hover to expand... Click to expand...
  11. CaffeinatedTech

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    I had wireshark running yesterday and left the VM window open so I could watch it. I had a call drop (using the 3CX soft phone via VoIP trunk), and nothing interesting came up in wireshark until I hung up. I could see all the invite and and ringing related messages, I could see where the call was connected but nothing happened when the line went silent. It's like 3CX just decides to stop transmitting audio.

    I was looking at other posts around the internet about people having clock-skew problems running 3CX on Hyper-V. I have my VM set to 2CPUs, but never have trouble with jittery IVR like I would with a linux based PBX. I tried setting the CPU to 1 this morning and gave it a bit of a test run by calling the voicemail system. I could see that it was spiking the CPU to 100% and jittering at the start of every ivr prompt and at the start of some messages. Figured that was because it was spiking, and set it back to 2CPUs.

    I want to get to the bottom of this, because I really don't want to have to run yet another dedicated machine, just to run a PBX that will have 2 simultaneous calls max.

    Do you think it needs more than 512MB of RAM?
     
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