Linksys SPA-3102 Disconnect the line periodicly.

Discussion in '3CX Phone System - General' started by vinzer, Sep 22, 2009.

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  1. vinzer

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    Hello all,

    Recently I added Linksys SPA-3102 for legacy phone line support here in the office. We have the following structure: there is a dedicated for 3CX Server software machine, version 7), there are few extensions on internal network (before the firewall) and there are few remote extensions which are in different parts of the city. At this moment we have good communication between all extensions (internal and external). After adding the SPA-3102 we got the access to our legacy phone line. It looks like it works well. At least when incomming call comes in it properly forward it to the Digital Receptionst, when we call outside using calling rule for this line it also works BUT:

    Sometimes, very often, it disconnects the call by it self. For example when I call from my extension thru legacy phone line (SPA-3102) after some 15-20 seconds it just drop the line and in the lof file it says Call terminated. Or when someone calls to legacy line from outside and after that transfer to the extension, especcially if it is remote extension, it also may drop connection. Below I am attaching the log from such call when someone called from outside to the legacy line, got to teh digital receptionist and after that choose some remote extension (by remote I mean different city in this case and no tunnel, just SIP phone on other end connected through our firewall to our server).

    --------------------------------------------------------------------
    22:46:03.625 [CM503008]: Call(51): Call is terminated

    22:46:03.609 [CM503008]: Call(51): Call is terminated

    22:46:00.515 [CM503007]: Call(51): Device joined: sip:317@74.59.33.148:5060

    22:46:00.515 [CM505001]: Ext.317: Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [Sipura/SPA841-3.1.1(a)] Transport: [sip:192.168.0.51:5060]

    22:46:00.515 [CM503002]: Call(51): Alerting sip:317@74.59.33.148:5060

    22:45:57.515 [CM503024]: Call(51): Calling Ext:Ext.317@[Dev:sip:317@74.59.33.148:5060]

    22:45:57.484 [CM503004]: Call(51): Route 1: Ext:Ext.317@[Dev:sip:317@74.59.33.148:5060]

    22:45:57.484 [CM503010]: Making route(s) to <sip:317@127.0.0.1:5060>

    22:45:52.765 [MS211000] C:51.1: 192.168.0.112:16414 is delivering DTMF using RTP payload (RFC2833). In-Band DTMF tone detection is disabled for this call segment.

    22:45:40.468 [CM503007]: Call(51): Device joined: sip:800@127.0.0.1:40600;rinstance=bb1c9907d7f349fc

    22:45:40.265 [CM503007]: Call(51): Device joined: sip:10000@192.168.0.112:5062

    22:45:40.265 [CM505001]: Ext.800: Device info: Device Identified: [Man: 3CX Ltd.;Mod: Voice Mail Menu;Rev: General] Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [3CX Voice Mail Menu] Transport: [sip:127.0.0.1:5060]

    22:45:40.265 [CM503002]: Call(51): Alerting sip:800@127.0.0.1:40600;rinstance=bb1c9907d7f349fc

    22:45:38.984 [CM503024]: Call(51): Calling Ext:Ext.800@[Dev:sip:800@127.0.0.1:40600;rinstance=bb1c9907d7f349fc]

    22:45:38.984 [CM503004]: Call(51): Route 1: Ext:Ext.800@[Dev:sip:800@127.0.0.1:40600;rinstance=bb1c9907d7f349fc]

    22:45:38.984 [CM503010]: Making route(s) to <sip:800@192.168.0.51:5060>

    22:45:38.984 [CM505002]: Gateway:[Videotron Gateway] Device info: Device Identified: [Man: Linksys;Mod: SPA Series;Rev: General] Capabilities:[reinvite, no-replaces, able-no-sdp, recvonly] UserAgent: [Linksys/SPA3102-5.1.7(GW)] Transport: [sip:192.168.0.51:5060]

    22:45:38.593 [CM503001]: Call(51): Incoming call from 10000@(Ln.10000@Videotron Gateway) to <sip:800@192.168.0.51:5060>

    22:45:38.593 [CM503012]: Inbound out-of-office hours rule (unnamed) for 10000 forwards to DN:800
    --------------------------------------------------------------------------------------------------------------------------------------


    I know that this problem could be discussed already but I could not find the answer WHY this happens. Please help me if you can.
    Thank you very much,
    Regards,
    Vlad
     
  2. leejor

    leejor Well-Known Member

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    Have a look at the the status page of the 3102 after a call drops, there is a field to show the reason for the last PSTN call disconnect. I found that some calls were too low in volume and i was getting a PSTN inactivity timeout after a given time, 3 minutes. You may want to try tweaking the PSTN disconnect detection settings under the PSTN tab, in fact you may want to disable all them completely and see if that makes a difference. If it does, then put them back as before, one at a time, until the problem reappears.
     
  3. vinzer

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    Thank you.
    My problem was in sensing of the voltage in phone line.
    I fixed all this by changing the settings in 3102 configuration
     
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