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Calls are intermittent

Discussion in '3CX Phone System - General' started by goagency, Sep 4, 2008.

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  1. goagency

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    When I call another extension, outbound or get a inbound call the audio is hit or miss. About 80% of the time the call is fine, but the other 20% of the time as soon as the call is picked up there is no audio. If I wait a few seconds I can sometime here some choppy audio but it never comes all the way in.

    I am using:
    3cx Free edition version 6.0.806.0 on windows server 2003
    A patton 4114 gateway
    SPA 962 and SPA 942 phones

    I have tried PBX delivers audio off and on, off seems to do better, buy still not fixed

    This has been really hard to trouble shoot as it only happens to 2 or three calls out of 10/

    Thank-you for your help
     
  2. amygoda

    amygoda Member

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    you have to find a reason to solve it
    ex: some times my remote ext. has one way audio
    I made a restart to each device and test
    I found the issue in my dsl router
     
  3. goagency

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    I am not sure what you mean, I want to solve it because clients call in all the time and can't hear us or we can't hear them, Or I can't hear the person at the front desk when I call them. I have reinstalled 3cx twice and still have the same results, and in the process restarted the router. I am only using 2 PTSN lines, and no remote extensions, but this issue is not always on incoming and outgoing calls through the Patton Gateway(I thought this was a issue with my Grandstream gateway, so I switched to the patton), I have issue when calling a local extension, but like I said not all the time, only about 80% of the time.

    Thanks
     
  4. Resolve

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    I hesitate to say this as it isn't really helpful....but we saw exactly the same problem on a site. It cleared up and we never found the cause - your description is actually uncanny.

    Anyway - some ideas which may or may not be relevant:

    Does it happen on internal calls ? What *internal* firewall if any are you using (XP firewall / Norton / AVG etc) ?

    How have you set up your network infrastructure - can you completely isolate phones from PC's (just for now - not an ongoing solution). If you've get any network issues and maybe a user or two net surfing to youtube or facebook for example, may magnify problems.

    Do you know how to use wireshark - and if so, what does it show (I significant massive jitter).

    Are there any POE devices in the setup ?

    What make is your router (do you even have an external VOIP account that you are using)?

    What is the spec of your machine (CPU/Memory) ?


    Tim
     
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  5. amygoda

    amygoda Member

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    see the firewall
    arrange the ports
    set the coddecs to g729 or 711
    can you a capture of a good call from the 3CX server and capture a bad call
    then compare and see the rtp
     
  6. goagency

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    THANK YOU FOR YOUR HELP this has been driving me crazy, almost to the point of quitting my job so I don't have to deal with it. Here is a Good call caputed:
    17:52:36.199 Call::Terminate [CM503008]: Call(10): Call is terminated
    17:52:36.090 LineCfg::getInboundTarget [CM503012]: Inbound out-of-office hours' rule for LN:704 forwards to DN:800
    17:52:36.090 LineCfg::getInboundTarget [CM503012]: Inbound out-of-office hours' rule for LN:704 forwards to DN:800
    17:52:36.090 LineCfg::getInboundTarget Looking for inbound target: called=anonymous; caller=**********
    17:52:36.074 LineCfg::getInboundTarget Looking for inbound target: called=anonymous; caller=**********
    17:52:36.074 Call::Terminate [CM503008]: Call(10): Call is terminated
    17:52:36.074 LineCfg::getInboundTarget [CM503012]: Inbound out-of-office hours' rule for LN:704 forwards to DN:800
    17:52:36.074 LineCfg::getInboundTarget Looking for inbound target: called=anonymous; caller=**********
    17:52:32.090 CallLeg::eek:nConfirmed Session 129 of leg C:10.1 is confirmed
    17:52:31.887 CallCtrl::eek:nLegConnected [CM503007]: Call(10): Device joined: sip:101@192.168.0.210:5060
    17:52:31.887 LineCfg::getInboundTarget [CM503012]: Inbound out-of-office hours' rule for LN:704 forwards to DN:800
    17:52:31.871 LineCfg::getInboundTarget Looking for inbound target: called=anonymous; caller=**********
    17:52:31.840 CallCtrl::eek:nLegConnected [CM503007]: Call(10): Device joined: sip:704@192.168.0.125:5060
    17:52:31.840 MediaServerReporting::SetRemoteParty [MS210007] C:10.1:Answer provided. Connection(by pass mode): 192.168.0.210:16430(16431)
    17:52:31.840 MediaServerReporting::SetRemoteParty [MS210001] C:10.2:Answer received. RTP connection: 192.168.0.210:16430(16431)
    17:52:31.840 Extension::printEndpointInfo [CM505001]: Ext.101: Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [] Transport: [sip:192.168.0.233:5060]
    17:52:31.840 CallLeg::setRemoteSdp Remote SDP is set for legC:10.2
    17:52:31.840 CallCtrl::eek:nAnsweredCall [CM503002]: Call(10): Alerting sip:101@192.168.0.210:5060
    17:52:29.418 MediaServerReporting::SetRemoteParty [MS210006] C:10.2:Offer provided. Connection(by pass mode): 192.168.0.125:4938(4939)
    17:52:29.387 CallCtrl::eek:nSelectRouteReq [CM503004]: Call(10): Calling: RingAll800:101Ext.101@[Dev:sip:101@192.168.0.210:5060]
    17:52:29.387 CallCtrl::eek:nSelectRouteReq [CM503010]: Making route(s) to [sip:800@sip.mydomain.com:5060]
    17:52:29.387 MediaServerReporting::SetRemoteParty [MS210000] C:10.1:Offer received. RTP connection: 192.168.0.125:4938(4939)
    17:52:29.387 CallLeg::setRemoteSdp Remote SDP is set for legC:10.1
    17:52:29.387 Line::printEndpointInfo [CM505002]: Gateway:[Pattonlines] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [Patton SN4114 JO EUI 00A0BA03F908 R5.2 2008-07-18 H323 SIP FXS FXO M5T SIP Stack/4.0.26.26] Transport: [sip:192.168.0.233:5060]
    17:52:29.387 LineCfg::getInboundTarget [CM503012]: Inbound out-of-office hours' rule for LN:704 forwards to DN:800
    17:52:29.371 LineCfg::getInboundTarget Looking for inbound target: called=anonymous; caller=**********
    17:52:29.293 CallCtrl::eek:nIncomingCall [CM503001]: Call(10): Incoming call from **********@(Ln.704@Pattonlines) to [sip:800@sip.mydomain.com:5060]
    17:52:29.293 LineCfg::getInboundTarget [CM503012]: Inbound out-of-office hours' rule for LN:704 forwards to DN:800
    17:52:29.277 LineCfg::getInboundTarget Looking for inbound target: called=anonymous; caller=**********
    17:52:29.277 CallLeg::eek:nNewCall [CM500002]: Info on incoming INVITE:
    INVITE sip:704@192.168.0.233 SIP/2.0
    Via: SIP/2.0/UDP 192.168.0.125:5060;branch=z9hG4bKef04087bef815f593
    Max-Forwards: 70
    Contact: [sip:15405207129@192.168.0.125:5060]
    To: [sip:anonymous@192.168.0.233]
    From: "VIRGINIA CALL"[sip:15405207129@192.168.0.125:5060];tag=d0740d86e0
    Call-ID: 788e8d80614a4daa
    CSeq: 20678 INVITE
    Proxy-Authorization: Digest username="704",realm="3CXPhoneSystem",nonce="12865038748:1dcb227ab494d4c3d4c287efdf9e8d94",uri="sip:704@192.168.0.233",response="3cece3bfcb4d904946326d64a661017f",algorithm=MD5
    Supported: replaces
    User-Agent: Patton SN4114 JO EUI 00A0BA03F908 R5.2 2008-07-18 H323 SIP FXS FXO M5T SIP Stack/4.0.26.26
    Content-Length: 0


    Here is a bad call:
    17:55:31.338 Call::Terminate [CM503008]: Call(17): Call is terminated
    17:55:31.292 LineCfg::getInboundTarget [CM503012]: Inbound out-of-office hours' rule for LN:704 forwards to DN:800
    17:55:31.292 LineCfg::getInboundTarget [CM503012]: Inbound out-of-office hours' rule for LN:704 forwards to DN:800
    17:55:31.292 LineCfg::getInboundTarget Looking for inbound target: called=anonymous; caller=**********
    17:55:31.276 LineCfg::getInboundTarget Looking for inbound target: called=anonymous; caller=**********
    17:55:31.276 Call::Terminate [CM503008]: Call(17): Call is terminated
    17:55:31.276 LineCfg::getInboundTarget [CM503012]: Inbound out-of-office hours' rule for LN:704 forwards to DN:800
    17:55:31.276 LineCfg::getInboundTarget Looking for inbound target: called=anonymous; caller=**********
    17:55:24.870 CallLeg::eek:nConfirmed Session 191 of leg C:17.1 is confirmed
    17:55:24.698 CallCtrl::eek:nLegConnected [CM503007]: Call(17): Device joined: sip:101@192.168.0.210:5060
    17:55:24.698 LineCfg::getInboundTarget [CM503012]: Inbound out-of-office hours' rule for LN:704 forwards to DN:800
    17:55:24.698 LineCfg::getInboundTarget Looking for inbound target: called=anonymous; caller=**********
    17:55:24.667 CallCtrl::eek:nLegConnected [CM503007]: Call(17): Device joined: sip:704@192.168.0.125:5060
    17:55:24.667 MediaServerReporting::SetRemoteParty [MS210001] C:17.2:Answer received. RTP connection: 192.168.0.210:16444(16445)
    17:55:24.667 Extension::printEndpointInfo [CM505001]: Ext.101: Device info: Device Identified: [Man: Linksys;Mod: SPA-941;Rev: General] Capabilities:[reinvite, no-replaces, able-no-sdp, recvonly] UserAgent: [Linksys/SPA962-5.2.8(SC)] Transport: [sip:192.168.0.233:5060]
    17:55:24.667 CallLeg::setRemoteSdp Remote SDP is set for legC:17.2
    17:55:24.667 CallCtrl::eek:nAnsweredCall [CM503002]: Call(17): Alerting sip:101@192.168.0.210:5060
    17:55:21.370 MediaServerReporting::SetRemoteParty [MS210002] C:17.2:Offer provided. Connection(transcoding mode): 192.168.0.233:7380(7381)
    17:55:21.339 CallCtrl::eek:nSelectRouteReq [CM503004]: Call(17): Calling: RingAll800:101Ext.101@[Dev:sip:101@192.168.0.210:5060]
    17:55:21.339 CallCtrl::eek:nSelectRouteReq [CM503010]: Making route(s) to [sip:800@sip.mydomain.com:5060]
    17:55:21.339 Line::printEndpointInfo [CM505002]: Gateway:[Pattonlines] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [Patton SN4114 JO EUI 00A0BA03F908 R5.2 2008-07-18 H323 SIP FXS FXO M5T SIP Stack/4.0.26.26] Transport: [sip:192.168.0.233:5060]
    17:55:21.339 LineCfg::getInboundTarget [CM503012]: Inbound out-of-office hours' rule for LN:704 forwards to DN:800
    17:55:21.339 LineCfg::getInboundTarget Looking for inbound target: called=anonymous; caller=15405207129
    17:55:21.245 MediaServerReporting::InitEndPoint [MS003005] C:17.1: Failed to create Endpoint: (destination=192.168.0.125)
    EndPoint: ID=00000C76@(LOCAL)
    LOGID=C:17.1 Status: MSEP_FAILED
    RTP:192.168.0.233:7378
    RTCP:192.168.0.233:7379
    STUN RTP:0.0.0.0:0
    STUN RTCP:0.0.0.0:0
    Coder:
    NOT SET
    101:telephony-event
    Party ptime:20
    Party RTP:0.0.0.0:0
    Party RTCP:0.0.0.0:0
    Decoders:
    [empty]
    17:55:21.245 MediaServerReporting::InitEndPoint [MS003003] C:17.1: RTCP socket 192.168.0.233:7379 binding failed with error code 10048
    17:55:21.245 MediaServerReporting::InitEndPoint [MS003002] C:17.1: RTP socket 192.168.0.233:7378 binding failed with error code 10048
    17:55:21.245 CallCtrl::eek:nIncomingCall [CM503001]: Call(17): Incoming call from **********@(Ln.704@Pattonlines) to [sip:800@sip.mydomain.com:5060]
    17:55:21.245 LineCfg::getInboundTarget [CM503012]: Inbound out-of-office hours' rule for LN:704 forwards to DN:800
    17:55:21.229 LineCfg::getInboundTarget Looking for inbound target: called=anonymous; caller=**********
    17:55:21.214 CallLeg::eek:nNewCall [CM500002]: Info on incoming INVITE:
    INVITE sip:704@192.168.0.233 SIP/2.0
    Via: SIP/2.0/UDP 192.168.0.125:5060;branch=z9hG4bK2cef40223cfb4432f
    Max-Forwards: 70
    Contact: [sip:15405207129@192.168.0.125:5060]
    To: [sip:anonymous@192.168.0.233]
    From: "VIRGINIA CALL"[sip:15405207129@192.168.0.125:5060];tag=f8b00e5952
    Call-ID: 44388a127c0e7127
    CSeq: 880 INVITE
    Proxy-Authorization: Digest username="704",realm="3CXPhoneSystem",nonce="12865038920:81cf2282cfc67913997daf7f560b94a7",uri="sip:704@192.168.0.233",response="f102dcf1f631f1c5c4e619dfdc340ae8",algorithm=MD5
    Supported: replaces
    User-Agent: Patton SN4114 JO EUI 00A0BA03F908 R5.2 2008-07-18 H323 SIP FXS FXO M5T SIP Stack/4.0.26.26
    Content-Length: 0


    Here is another bad call, (sometimes if I watch the log, the log will change from what is below to something that looks like the bad call above)

    17:51:48.840 Call::Terminate [CM503008]: Call(9): Call is terminated
    17:51:48.824 LineCfg::getInboundTarget [CM503012]: Inbound out-of-office hours' rule for LN:704 forwards to DN:800
    17:51:48.824 LineCfg::getInboundTarget [CM503012]: Inbound out-of-office hours' rule for LN:704 forwards to DN:800
    17:51:48.809 LineCfg::getInboundTarget Looking for inbound target: called=anonymous; caller=**********
    17:51:48.809 LineCfg::getInboundTarget Looking for inbound target: called=anonymous; caller=**********
    17:51:48.809 Call::Terminate [CM503008]: Call(9): Call is terminated
    17:51:48.809 LineCfg::getInboundTarget [CM503012]: Inbound out-of-office hours' rule for LN:704 forwards to DN:800
    17:51:48.809 LineCfg::getInboundTarget Looking for inbound target: called=anonymous; caller=**********
    17:51:42.699 CallLeg::eek:nConfirmed Session 119 of leg C:9.1 is confirmed
    17:51:42.559 CallCtrl::eek:nLegConnected [CM503007]: Call(9): Device joined: sip:101@192.168.0.210:5060
    17:51:42.559 LineCfg::getInboundTarget [CM503012]: Inbound out-of-office hours' rule for LN:704 forwards to DN:800
    17:51:42.559 LineCfg::getInboundTarget Looking for inbound target: called=anonymous; caller=**********
    17:51:42.481 CallCtrl::eek:nLegConnected [CM503007]: Call(9): Device joined: sip:704@192.168.0.125:5060
    17:51:42.481 MediaServerReporting::SetRemoteParty [MS210003] C:9.1:Answer provided. Connection(transcoding mode):192.168.0.233:7346(7347)
    17:51:42.481 Extension::printEndpointInfo [CM505001]: Ext.101: Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [] Transport: [sip:192.168.0.233:5060]
    17:51:42.481 CallCtrl::eek:nAnsweredCall [CM503002]: Call(9): Alerting sip:101@192.168.0.210:5060
    17:51:31.231 Call::Terminate [CM503008]: Call(8): Call is terminated


    Firewall: I was using windows firewall, but it has been disabled.

    I have not separated the phones from the computers, the phones have a built in switch and the computers connect through the phones. I will look into wire shark.

    Router: I am using a linksys 4000 with no VOIP

    Server Specs
    Processor: Dual-Core Intel Xeon Processor 5120 (1.86 GHz, 1066 FSB)
    Memory: 3gb

    Thanks
     
  7. Resolve

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    Ok - This looks to me like a codec problem, and on one site I had a similar problem with Linksys phones. We got round that one by forcing the phones to use G711u

    Have you made any changes to codec choices on any of the phones/3cx or for that matter on the Patton ?

    Do you know how to use wireshark - it would be good to see the negotiation take place. Secondly, and I appreciate from experience this is difficult to do in a "live" setup, but have you tried using the Voipclient/headset - and if so does the problem still recur ?


    Another thought - is there any way, even temporarily you can eliminate the phone switch(s) ? I've seen a similar(not the same though) problem due to a dodgy patch cable ?



    Tim
     
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  8. KR_Monterroyo

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    We are getting choppy lines when we are the one doing the outbound call. We are located in the Philippines and our main office is in Singapore. When customers call in, we do not have a problem but we are the one who does the outgoing call, the customers can't hear us although we can hear the customer very well. When we call our hotline, both lines are choppy. Please advise asap
     
  9. lneblett

    lneblett Well-Known Member

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    OK, a little confused as your message indicates a number of different scenarios. We need to narrow down the issues to specifics as you indicated:

    1. When I call another extension, outbound or get a inbound call the audio is hit or miss.
    2. clients call in all the time and can't hear us or we can't hear them, Or I can't hear the person at the front desk when I call them
    3. I am only using 2 PTSN lines, and no remote extensions
    4. I have issue when calling a local extension, but like I said not all the time, only about 80% of the time

    The first thing is that you are using a woefully outdated version of 3CX (v6?). Is this a new install?????? You should upgrade to at least version 12 at a minimum and preferably 12.5, but I am uncertain if these are supported by WinServ2003.
    Second, do not use G729 as that codec is likely not supported in a free version and even if so, limited to half of the available call volume of the other available codecs as this is still covered by a patent and requires royalty fees. Use g711a or g711u. Make sure that whichever codec you select it is set to be the primary codec at the Patton, 3CX and the phones.
    Third, the phones you have elected to use are not supported by 3CX, but if compliant to SIP, should work, but the feature sets may not all function correctly. Nevertheless, you may want to ensure that these as well as the Patton gateway are updated to the latest firmware.

    If I understand correctly, all calls are either coming in or going out thru the Patton or, are internal going from extension to extension.

    Once you have done the above, then you are in a position to better identify and fix the issues.

    Choppy or no audio is an indication that the RTP (voice) streams are running into issues, but the hardware and installation all need to be at a level where we can assume that by being updated that they are not contributing to the issues...or at least not so much as maybe the case where you are now.
     
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