Grandstream 286 lost reg and didn't hang up

Discussion in '3CX Phone System - General' started by craigreilly, Sep 6, 2012.

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  1. craigreilly

    craigreilly Well-Known Member

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    Hope you can point me in the right direction on what might cause this.
    Extension 4004 is a Grandstream 286 and connected to it is a Polycom Analog Speakerphone.
    Out trunk is a Patton 4940 connected to a PRI

    1) Ext 4004 made a long distance call at 11:57:52 am
    2) At about 12:14 the grandstream unregistered.
    3) The call ended around 12:40 pm – but no sign of it in the log. The phone has dialtone but can not make calls as it is not registered.
    a. I see the call still going in MyPhone at 1:12pm so I unplug the Grandstream and cycle the power
    b. The call continues on the Patton (I can see it under Port/Trunk Status)
    4) The only way I could see to end the call, was to login to MyPhone as that extension and terminate it that way.

    This is a little concerning because that’s about 30 minutes of LD Charges that were used for no reason at all.

    The next line in the log is at 13:14:04 when I cycled the power on the grandstream and it reregistered.

    1) Is there another way to terminate a call on the trunk ?
    2) Is there a setting on the Grandstream that should be changed so reguistrations take place more often or something?



    06-09-2012 12:14:44.943 Endpoint Extn:4004 has removed contact <sip:4004@10.0.3.102:5060>
    06-09-2012 12:14:44.943 [CM504002]: Endpoint Extn:4004: a contact is unregistered. Contact(s): []
    06-09-2012 11:57:54.658 Session 3227508 of leg L:9327.1[Extn] is confirmed
    06-09-2012 11:57:54.658 L:9327.1[Extn] got Confirmed Confirmed Recv Req ACK from 10.0.3.102:5060 tid=f33e451e357539d6 Call-ID=e20fa375da2148e5@10.0.3.102:
    ACK sip:4075558606@10.0.0.12:5060 SIP/2.0
    Via: SIP/2.0/UDP 10.0.3.102;branch=z9hG4bKf33e451e357539d6
    Max-Forwards: 70
    Contact: <sip:4004@10.0.3.102;user=phone>
    To: <sip:4075558606@10.0.0.12;user=phone>;tag=5b67da04
    From: "Conference Room"<sip:4004@10.0.0.12;user=phone>;tag=553ff24ebdb70626
    Call-ID: e20fa375da2148e5@10.0.3.102
    CSeq: 55862 ACK
    Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, INFO, SUBSCRIBE, UPDATE
    Proxy-Authorization: Digest username="4004", realm="3CXPhoneSystem", algorithm=MD5, uri="sip:4075558606@10.0.0.12:5060", nonce="414d535c0659833070:3982c74f4639ebf317a09cb0090dfb64", response="0a28ab18fe564625f632fa1f93be498f"
    User-Agent: Grandstream HT287 1.1.0.45 DevId 000b823074d8
    Content-Length: 0
    06-09-2012 11:57:54.608 [CM503007]: Call(C:9327): Extn:4004 has joined, contact <sip:4004@10.0.3.102:5060>
    06-09-2012 11:57:54.608 Session 3227508 of leg L:9327.1[Extn] is connected
    06-09-2012 11:57:54.608 L:9327.1[Extn] got Connected Connected Send 200/INVITE from 0.0.0.0:0 tid=05bc02cc7a2dbae5 Call-ID=e20fa375da2148e5@10.0.3.102:
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 10.0.3.102;branch=z9hG4bK05bc02cc7a2dbae5
    Contact: <sip:4075558606@10.0.0.12:5060>
    To: <sip:4075558606@10.0.0.12;user=phone>;tag=5b67da04
    From: "Conference Room"<sip:4004@10.0.0.12;user=phone>;tag=553ff24ebdb70626
    Call-ID: e20fa375da2148e5@10.0.3.102
    CSeq: 55862 INVITE
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
    Content-Type: application/sdp
    Supported: replaces
    User-Agent: 3CXPhoneSystem 11.0.25940.0
    Content-Length: 231

    v=0
    o=3cxPS 202517774336 27212644353 IN IP4 10.0.0.12
    s=3cxPS Audio call
    c=IN IP4 10.0.3.1
    t=0 0
    m=audio 5222 RTP/AVP 0 8 18
    a=rtpmap:0 PCMU/8000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=no
    a=sendrecv
    06-09-2012 11:57:54.608 L:9327.1[Extn] Sending: OnSendResp Send 200/INVITE from 0.0.0.0:0 tid=05bc02cc7a2dbae5 Call-ID=e20fa375da2148e5@10.0.3.102:
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 10.0.3.102;branch=z9hG4bK05bc02cc7a2dbae5
    Contact: <sip:4075558606@10.0.0.12:5060>
    To: <sip:4075558606@10.0.0.12;user=phone>;tag=5b67da04
    From: "Conference Room"<sip:4004@10.0.0.12;user=phone>;tag=553ff24ebdb70626
    Call-ID: e20fa375da2148e5@10.0.3.102
    CSeq: 55862 INVITE
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
    Content-Type: application/sdp
    Supported: replaces
    Content-Length: 231

    v=0
    o=3cxPS 202517774336 27212644353 IN IP4 10.0.0.12
    s=3cxPS Audio call
    c=IN IP4 10.0.3.1
    t=0 0
    m=audio 5222 RTP/AVP 0 8 18
    a=rtpmap:0 PCMU/8000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=no
    a=sendrecv
    06-09-2012 11:57:54.608 L:9327.1[Extn]: Terminating targets, reason: SIP ;cause=200 ;text="Call completed elsewhere"
    06-09-2012 11:57:52.968 L:9327.1[Extn] Sending: OnSendResp Send 180/INVITE from 0.0.0.0:0 tid=05bc02cc7a2dbae5 Call-ID=e20fa375da2148e5@10.0.3.102:
    SIP/2.0 180 Ringing
    Via: SIP/2.0/UDP 10.0.3.102;branch=z9hG4bK05bc02cc7a2dbae5
    Contact: <sip:4075558606@10.0.0.12;user=phone>
    To: <sip:4075558606@10.0.0.12;user=phone>;tag=5b67da04
    From: "Conference Room"<sip:4004@10.0.0.12;user=phone>;tag=553ff24ebdb70626
    Call-ID: e20fa375da2148e5@10.0.3.102
    CSeq: 55862 INVITE
    Content-Length: 0
    06-09-2012 11:57:52.821 [Flow] Call(C:9327): making call from L:9327.1[Extn] to T:Line:10003>>14075558606@[Dev:sip:10003@10.0.3.1:5060]
    06-09-2012 11:57:52.821 [CM503027]: Call(C:9327): From: Extn:4004 ("Conference Room" <sip:4004@10.0.0.12:5060>) to T:Line:10003>>14075558606@[Dev:sip:10003@10.0.3.1:5060]
    06-09-2012 11:57:52.821 [CM503010]: Call(C:9327): Making route(s) from Extn:4004 to <sip:4075558606@10.0.0.12:5060>
    06-09-2012 11:57:52.821 Remote SDP is set for leg L:9327.1[Extn]
    06-09-2012 11:57:52.821 [CM505001]: Endpoint Extn:4004: Device info: Device Identified: [Man: Yealink;Mod: T28;Rev: General] Capabilities:[reinvite, replaces, unable-no-sdp, no-recvonly] UserAgent: [Grandstream HT287 1.1.0.45 DevId 000b823074d8] PBX contact: [sip:4004@10.0.0.12:5060]
    06-09-2012 11:57:52.820 [CM500002]: Call(C:9327): Info on incoming INVITE from Extn:4004:
    Invite-IN Recv Req INVITE from 10.0.3.102:5060 tid=05bc02cc7a2dbae5 Call-ID=e20fa375da2148e5@10.0.3.102:
    INVITE sip:4075558606@10.0.0.12;user=phone SIP/2.0
    Via: SIP/2.0/UDP 10.0.3.102;branch=z9hG4bK05bc02cc7a2dbae5
    Max-Forwards: 70
    Contact: <sip:4004@10.0.3.102;user=phone>
    To: <sip:4075558606@10.0.0.12;user=phone>
    From: "Conference Room"<sip:4004@10.0.0.12;user=phone>;tag=553ff24ebdb70626
    Call-ID: e20fa375da2148e5@10.0.3.102
    CSeq: 55862 INVITE
    Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, INFO, SUBSCRIBE, UPDATE
    Content-Type: application/sdp
    Proxy-Authorization: Digest username="4004",realm="3CXPhoneSystem",algorithm=MD5,uri="sip:4075558606@10.0.0.12;user=phone",nonce="414d535c0659833070:3982c74f4639ebf317a09cb0090dfb64",response="9429125928c533ae541ee16aa1d8d29b"
    Supported: replaces, timer
    User-Agent: Grandstream HT287 1.1.0.45 DevId 000b823074d8
    Content-Length: 318

    v=0
    o=4004 8000 8001 IN IP4 10.0.3.102
    s=SIP Call
    c=IN IP4 10.0.3.102
    t=0 0
    m=audio 5004 RTP/AVP 0 8 4 18 2 97
    a=sendrecv
    a=rtpmap:0 PCMU/8000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:4 G723/8000
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=no
    a=rtpmap:2 G726-32/8000
    a=rtpmap:97 iLBC/8000
    a=fmtp:97 mode=20
    a=ptime:20
    06-09-2012 11:57:52.820 [CM503001]: Call(C:9327): Incoming call from Extn:4004 to sip:4075558606@10.0.0.12:5060
     
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  2. lneblett

    lneblett Well-Known Member

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    To force a disconnect, go to the Ports/Trunk Status tab and then look just above and you should see a "Disconnect Call" button.

    The 286/287 has been around for some time and is generally perceived as a workhorse. I am uncertain how you have it configured, but here are some items that may help:

    1. Allow outgoing calls without registration = no
    2. Disable voice prompt = no
    3. Registration expiration - to whatever you want, the default is normaly 3600 sec (1 hour).
    4. Session Expiration - set to whatever, default is 180 (reivite set to on with the extension of interest in 3CX)
    6. Unregister on reboot = yes

    Can you check and compare and let me know which may have been different? We can then go from there....maybe.
     
  3. craigreilly

    craigreilly Well-Known Member

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    There's the button - I tried the ol' Right Click. How I didn't see that I do not know.
    I'll try the tips. Thanks!





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  4. craigreilly

    craigreilly Well-Known Member

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    1. Allow outgoing calls without registration
    = no -> UPDATED
    2. Disable voice prompt = no
    = no already
    3. Registration expiration - to whatever you want, the default is normaly 3600 sec (1 hour).
    = 3600 already
    4. Session Expiration - set to whatever, default is 180 (reivite set to on with the extension of interest in 3CX)
    = 180 already and reinvite ON already
    6. Unregister on reboot = yes
    = yes already

    So - I only changed option 1.
     
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