dropping mobile calls after a couple of minutes

Discussion in '3CX Phone System - General' started by gsaccountants, Apr 16, 2010.

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  1. gsaccountants

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    Hi,
    i have (another!) problem but when making calls from internal extensions to mobile's. not all mobiles but some - everything seems fine and then all of sudden I just get a dead tone??
    is there any reason why that might happen?

    this is the 3cx log from last time. i am using a linksys3102 on 192.168.1.100 and the pbx is 192.168.1.6. I tried transferring the calls between extensions but that doesn't seem related.

    Code:
    14:32:36.283  Active calls counted toward license limit: []
    14:32:04.283  Active calls counted toward license limit: []
    14:31:32.283  Active calls counted toward license limit: []
    14:31:00.284  Active calls counted toward license limit: []
    14:30:28.284  Active calls counted toward license limit: []
    14:30:24.409  [MS105000] C:20.1: No RTP packets were received:remoteAddr=192.168.1.100:16412,extAddr=0.0.0.0:0,localAddr=192.168.1.6:7072
    14:30:23.237  [CM503008]: Call(20): Call is terminated
    14:30:07.237  [CM503007]: Call(20): Device joined: sip:101@192.168.1.19:5060;transport=UDP
    14:30:07.237  [CM505001]: Ext.101: Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [Mitel-5235-SIP-Phone 07.02.00.04 08000F29815D] PBX contact: [sip:101@192.168.1.6:5060]
    14:30:07.237  [CM503002]: Call(20): Alerting sip:101@192.168.1.19:5060;transport=UDP
    14:29:58.534  [CM503025]: Call(20): Calling Ext:Ext.101@[Dev:sip:101@192.168.1.19:5060;transport=UDP]
    14:29:58.471  [CM503004]: Call(20): Route 1: Ext:Ext.101@[Dev:sip:101@192.168.1.19:5060;transport=UDP]
    14:29:58.471  [CM503010]: Making route(s) to <sip:101@192.168.1.6>
    14:29:56.284  Active calls counted toward license limit: [20]
    14:29:29.409  [CM503007]: Call(20): Device joined: sip:102@192.168.1.20:5060;transport=UDP
    14:29:29.409  [CM505001]: Ext.102: Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [Mitel-5235-SIP-Phone 07.02.00.04 08000F297EC8] PBX contact: [sip:102@192.168.1.6:5060]
    14:29:29.409  [CM503002]: Call(20): Alerting sip:102@192.168.1.20:5060;transport=UDP
    14:29:24.284  Active calls counted toward license limit: [20]
    14:29:21.769  [CM503025]: Call(20): Calling Ext:Ext.102@[Dev:sip:102@192.168.1.20:5060;transport=UDP]
    14:29:21.706  [CM503004]: Call(20): Route 1: Ext:Ext.102@[Dev:sip:102@192.168.1.20:5060;transport=UDP]
    14:29:21.706  [CM503010]: Making route(s) to <sip:102@192.168.1.6>
    14:28:52.284  Active calls counted toward license limit: [20]
    14:28:20.285  Active calls counted toward license limit: [20]
    14:27:48.285  Active calls counted toward license limit: [20]
    14:27:16.285  Active calls counted toward license limit: [20]
    14:26:44.285  Active calls counted toward license limit: [20]
    14:26:12.285  Active calls counted toward license limit: [20]
    14:25:40.286  Active calls counted toward license limit: [20]
    14:25:33.786  [CM503003]: Call(20): Call to sip:201@87.244.76.5 has failed; Cause: 487 Request Terminated; from IP:82.112.156.220:5060
    14:25:33.739  [CM503003]: Call(20): Call to sip:102@192.168.1.6 has failed; Cause: 487 Request Terminated; from IP:192.168.1.20:5060
    14:25:33.739  [CM503003]: Call(20): Call to sip:100@192.168.1.6 has failed; Cause: 487 Request Terminated; from IP:192.168.1.21:5060
    14:25:33.583  [CM503007]: Call(20): Device joined: sip:101@192.168.1.19:5060;transport=UDP
    14:25:33.583  [CM503007]: Call(20): Device joined: sip:10002@192.168.1.100:5062
    14:25:33.583  [CM505001]: Ext.101: Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [Mitel-5235-SIP-Phone 07.02.00.04 08000F29815D] PBX contact: [sip:101@192.168.1.6:5060]
    14:25:33.583  [CM503002]: Call(20): Alerting sip:101@192.168.1.19:5060;transport=UDP
    14:25:30.426  [CM503025]: Call(20): Calling RingAll800:100Ext.100101Ext.101102Ext.102105Ext.105201Ext.201202Ext.202@[Dev:sip:201@82.112.156.220:5060;transport=UDP]
    14:25:30.426  [CM503025]: Call(20): Calling RingAll800:100Ext.100101Ext.101102Ext.102105Ext.105201Ext.201202Ext.202@[Dev:sip:201@10.245.114.63:49723;rinstance=3536E624260EED24E6FB]
    14:25:30.411  [CM503025]: Call(20): Calling RingAll800:100Ext.100101Ext.101102Ext.102105Ext.105201Ext.201202Ext.202@[Dev:sip:102@192.168.1.20:5060;transport=UDP]
    14:25:30.411  [CM503025]: Call(20): Calling RingAll800:100Ext.100101Ext.101102Ext.102105Ext.105201Ext.201202Ext.202@[Dev:sip:101@192.168.1.19:5060;transport=UDP]
    14:25:30.411  [CM503025]: Call(20): Calling RingAll800:100Ext.100101Ext.101102Ext.102105Ext.105201Ext.201202Ext.202@[Dev:sip:100@192.168.1.21:5060;transport=UDP]
    14:25:30.364  [CM503004]: Call(20): Route 1: RingAll800:100Ext.100101Ext.101102Ext.102105Ext.105201Ext.201202Ext.202@[Dev:sip:100@192.168.1.21:5060;transport=UDP,Dev:sip:101@192.168.1.19:5060;transport=UDP,Dev:sip:102@192.168.1.20:5060;transport=UDP,Dev:sip:201@10.245.114.63:49723;rinstance=3536E624260EED24E6FB,Dev:sip:201@82.112.156.220:5060;transport=UDP]
    14:25:30.364  [CM503010]: Making route(s) to <sip:800@192.168.1.6:5060>
    14:25:30.348  [CM505002]: Gateway:[Linksys SPA3102] Device info: Device Identified: [Man: Linksys;Mod: SPA Series;Rev: General] Capabilities:[reinvite, no-replaces, able-no-sdp, recvonly] UserAgent: [Linksys/SPA3102-5.1.10(GW)] PBX contact: [sip:10002@192.168.1.6:5060]
    14:25:30.348  [CM503001]: Call(20): Incoming call from 07700727430@(Ln.10002@Linksys SPA3102) to <sip:800@192.168.1.6:5060>
    14:25:30.348  [CM503012]: Inbound out-of-office hours rule (unnamed) for 10002 forwards to DN:800
    14:25:08.286  Active calls counted toward license limit: []
    14:24:36.286  Active calls counted toward license limit: []
    14:24:32.114  [CM503008]: Call(19): Call is terminated
    14:24:19.333  [CM503007]: Call(19): Device joined: sip:10002@192.168.1.100:5062
    14:24:19.333  [CM503007]: Call(19): Device joined: sip:101@192.168.1.19:5060;transport=UDP
    14:24:19.333  [CM505002]: Gateway:[Linksys SPA3102] Device info: Device Identified: [Man: Linksys;Mod: SPA Series;Rev: General] Capabilities:[reinvite, no-replaces, able-no-sdp, recvonly] UserAgent: [Linksys/SPA3102-5.1.10(GW)] PBX contact: [sip:10002@192.168.1.6:5060]
    14:24:19.317  [CM503002]: Call(19): Alerting sip:10002@192.168.1.100:5062
    14:24:19.255  [CM503025]: Call(19): Calling PSTNline:07700727430@(Ln.10002@Linksys SPA3102)@[Dev:sip:10002@192.168.1.100:5062]
    14:24:19.208  [CM503004]: Call(19): Route 1: PSTNline:07700727430@(Ln.10002@Linksys SPA3102)@[Dev:sip:10002@192.168.1.100:5062]
    14:24:19.208  [CM503010]: Making route(s) to <sip:07700727430@192.168.1.6>
    14:24:19.192  [CM505001]: Ext.101: Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [Mitel-5235-SIP-Phone 07.02.00.04 08000F29815D] PBX contact: [sip:101@192.168.1.6:5060]
    14:24:19.192  [CM503001]: Call(19): Incoming call from Ext.101 to <sip:07700727430@192.168.1.6>
    14:24:04.286  Active calls counted toward license limit: []
    14:23:32.286  Active calls counted toward license limit: []
    14:23:00.287  Active calls counted toward license limit: []
    14:22:28.287  Active calls counted toward license limit: []
    14:21:56.287  Active calls counted toward license limit: []
    I will owe beer to those that can help me!
    thanks,
    Alun
     
  2. gsaccountants

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    sorry maybe i should clarify but I everything is normal, calls connect fine, call quality is fine, everything is just great for 2-3 minutes then gone!
     
  3. mfm

    mfm Active Member

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    Hi,

    You are using an unsupported gateway, I would start by contacting mittel, but I would use a supported device that has been tested and is known to work well with 3CX you will also find more people able to help as they are more commonly used.
     
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  4. gsaccountants

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    hi thanks. the gateway is a linksys spa3102 which is supported.
    al.
     
  5. mfm

    mfm Active Member

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    Apologies, to take this further we are going to have to need to take a look at a wireshark capture. You mentioned this does not happen on all mobiles, is it to particular mobiles or just randomly on all mobiles?
     
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  6. leejor

    leejor Well-Known Member

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    On the 3102, after a call has ended, you can check the status page and it will tell you the reason for the disconnect. One problem I was having was with low level (generally) long distance calls dropping after a few minutes. If you find that "inactivity" is the reason for the dropped call, you may want to play with the settings in the PSTN Line tab/PSTN Disconnection Detection section. You can try disabling or extending the Long Silence Duration. See if that changes anything.
     
  7. gsaccountants

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    well it seems to happen on some mobiles, doesn't seem to be network, but often with a call to a blackberry!


    the reason for the call drop just says "Voip call ended" so not a lot of help there unfortunately. I had tried playing around with the various disconnect options including turning off CPC which I thought helped but it happened again today.

    Al.
     
  8. gsaccountants

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    Code:
    Hook State:	On	Line Voltage:	-50 (V)
    Loop Current:	0.0 (mA)	Registration State:	Registered
    Last Registration At:	1/12/2003 07:46:15	Next Registration In:	1399 s
    Last Called VoIP Number:	10002	Last Called PSTN Number:	01534440318
    Last VoIP Caller:	10002	Last PSTN Caller:	, 07797810591
    Last PSTN Disconnect Reason:	VoIP Call Ended	PSTN Activity Timer:	30000 (ms
    
    not strictly code but all the info i have on the disconnects...
     
  9. gsaccountants

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    is there a way to get at the linksys logs? nothing seems that straightforward!
     
  10. Cjay

    Cjay New Member

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    You have got the most intelligible output from the SPA-3102 already. You could set up a syslog server to capture the diagnostic info thrown out by the 3102, but honestly this is unlikely to help you further.

    The 'Last PSTN Disconnect reason' is the most useful: VoIP call ended means the disconnect was initiated by 3cx rather than triggered by something on the PSTN network (though as a matter of interest I have detect PSTN long silence & VoIP long silence set to 'NO' as well as Detect disconnect set to 'NO'. I find CPC works for me [BT in the UK]).

    I notice you are running SPA-3102 firmware 5.1.10 - I have seen some reports of bugs in this firmware which currently can only be fixed by going back to v 5.1.7 If all else fails you might want to try this - it has been rock solid for me!

    Good luck,
    Chris
     
  11. SY

    SY Well-Known Member
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    Excuse me, but logs on 3CX PBX side are always available.
    Use "help"/"generate support information" to collect it. (I'm talking about menu available in PBX management console)

    Thanks
     
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  12. leejor

    leejor Well-Known Member

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    Does it always happen to the same numbers/provider or will it stay up to a particular number one time but drop to the same number another time? As you are going through the 3102, other than low level or CPC disconnect there should not be any difference for a call going to one number or another. You did disable the Detect Long Silence option, right?

    Since the 3102 is showing that the VoIP end is what dropped I would...

    Look at the 3CX (verbatim mode) logs, see if it gives you more info on what is ending the call. Try the calls from another type of set, the 3CX softphone for example, to see if the set(s) is ending the call. Wireshark on a hub connected between the set and 3CX should confirm whether the set is or isn't the cause.
     
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