Auto attendant interfering with current call

Discussion in '3CX Phone System - General' started by rodster, Jan 20, 2010.

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  1. rodster

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    This is a strange behavior in one of our installation. The problem can be replicated.

    Here's the setup:
    -trunks are connected to SPA 3102
    -extensions are connected to SPA 2102
    -the basics works; can call-in, can call-out, extensions can call other extensions, voicemail is ok, etc

    Here's the problem:
    1. an incoming call is received then an auto-attendant is played to the caller, and if no keys are pressed the call is transferred to an extension (works as configured)
    2. the call is then answered by the extension where the incoming call is transferred to.
    3. while that call is on, another extension tries to make an outside call. However, since the trunk line is still being used by the incoming call as described in #1, this other extension gets a message saying that the call cannot proceed because there is no trunk available. This is expected and all is good up to this point.
    4. however, the extension who received the incoming call in #1 and while still on the the incoming call, the auto attendent is playing on the line. Since the call is still on going, the caller on the trunk and the extension that accepted the call hears the auto attendant playing in the line.

    The auto attendent interferes, via playing while a call is established, with the current call only if there's another extension trying to make an unsuccessful outside call.

    It's an annoying problem and we can't seem to solve. Help please?

    Here's the call logs:

    18:14:36.256 [CM503008]: Call(1): Call is terminated
    18:14:21.365 [CM503008]: Call(3): Call is terminated
    18:14:21.162 Session 81 of leg C:4.1 is confirmed
    18:14:21.084 [CM503007]: Call(4): Device joined: sip:800@127.0.0.1:40600;rinstance=89cc8c9ff374de72
    18:14:21.069 [CM503007]: Call(4): Device joined: sip:10000@192.168.0.20:5061
    18:14:21.069 [MS210005] C:4.1:Answer provided. Connection(proxy mode):192.168.0.247:7018(7019)
    18:14:21.053 [MS210001] C:4.2:Answer received. RTP connection[unsecure]: 127.0.0.1:40616(40617)
    18:14:21.053 Remote SDP is set for legC:4.2
    18:14:21.053 [CM505001]: Ext.800: Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [3CX IVR] PBX contact: [sip:800@127.0.0.1:5060]
    18:14:21.053 [CM503002]: Call(4): Alerting sip:800@127.0.0.1:40600;rinstance=89cc8c9ff374de72
    18:14:20.881 [CM503025]: Call(4): Calling Ext:Ext.800@[Dev:sip:800@127.0.0.1:40600;rinstance=89cc8c9ff374de72]
    18:14:20.881 [MS210004] C:4.2:Offer provided. Connection(proxy mode): 127.0.0.1:7020(7021)
    18:14:20.866 [CM503004]: Call(4): Route 1: Ext:Ext.800@[Dev:sip:800@127.0.0.1:40600;rinstance=89cc8c9ff374de72]
    18:14:20.866 [CM503010]: Making route(s) to <sip:800@192.168.0.247:5060>
    18:14:20.866 [MS210000] C:4.1:Offer received. RTP connection: 192.168.0.20:16464(16465)
    18:14:20.850 Remote SDP is set for legC:4.1
    18:14:20.850 [CM505002]: Gateway:[2213167] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, no-replaces, able-no-sdp, recvonly] UserAgent: [Linksys/SPA3102-3.3.6(GW)] PBX contact: [sip:10000@192.168.0.247:5060]
    18:14:20.834 [CM503001]: Call(4): Incoming call from 10000@(Ln.10000@2213167) to <sip:800@192.168.0.247:5060>
    18:14:20.834 [CM503012]: Inbound out-of-office hours rule (unnamed) for 10000 forwards to DN:800
    18:14:20.819 Looking for inbound target: called=10000; caller=10000
    18:14:20.819 [CM500002]: Info on incoming INVITE:
    INVITE sip:10000@192.168.0.247 SIP/2.0
    Via: SIP/2.0/UDP 192.168.0.20:5061;branch=z9hG4bK-ca6cff10
    Max-Forwards: 70
    Contact: "2213167"<sip:10000@192.168.0.20:5061>
    To: <sip:10000@192.168.0.247>
    From: "2213167"<sip:10000@192.168.0.247>;tag=3b1ef830ce5ff83co1
    Call-ID: cfffde40-a43dc96c@192.168.0.20
    CSeq: 102 INVITE
    Expires: 240
    Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
    Proxy-Authorization: Digest username="10000",realm="3CXPhoneSystem",nonce="414d535c0167687c13:e921700811dab542523eb10e267bfa1c",uri="sip:10000@192.168.0.247",algorithm=MD5,response="79443e44bcceedc863fa08498fb2181d"
    Supported: x-sipura
    User-Agent: Linksys/SPA3102-3.3.6(GW)
    Content-Length: 0
    Remote-Party-ID: 2213167 <sip:10000@192.168.0.247>;screen=yes;party=calling

    18:14:15.106 Session 68 of leg C:3.1 is confirmed
    18:14:15.012 [CM503007]: Call(3): Device joined: sip:EndCall@127.0.0.1:40600;rinstance=26223c814f7ae7d6
    18:14:14.997 [CM503007]: Call(3): Device joined: sip:203@192.168.0.20:5060
    18:14:14.981 [MS210005] C:3.1:Answer provided. Connection(proxy mode):192.168.0.247:7012(7013)
    18:14:14.981 [MS210001] C:3.3:Answer received. RTP connection[unsecure]: 127.0.0.1:40614(40615)
    18:14:14.981 Remote SDP is set for legC:3.3
    18:14:14.981 [CM503002]: Call(3): Alerting sip:EndCall@127.0.0.1:40600;rinstance=26223c814f7ae7d6
    18:14:14.841 [CM503025]: Call(3): Calling Unknown:Ext.EndCall@[Dev:sip:EndCall@127.0.0.1:40600;rinstance=26223c814f7ae7d6]
    18:14:14.841 [MS210004] C:3.3:Offer provided. Connection(proxy mode): 127.0.0.1:7016(7017)
    18:14:14.778 [CM503016]: Call(3): Attempt to reach <sip:92211234@192.168.0.247> failed. Reason: Server Failure
    18:14:14.762 [CM503003]: Call(3): Call to sip:2211234@192.168.0.20:5061 has failed; Cause: 503 Service Unavailable; from IP:192.168.0.20:5061
    18:14:14.731 [CM503025]: Call(3): Calling PSTNline:2211234@(Ln.10000@2213167)@[Dev:sip:10000@192.168.0.20:5061]
    18:14:14.716 [MS210006] C:3.2:Offer provided. Connection(by pass mode): 192.168.0.20:16462(16463)
    18:14:14.700 [CM503004]: Call(3): Route 1: PSTNline:2211234@(Ln.10000@2213167)@[Dev:sip:10000@192.168.0.20:5061]
    18:14:14.684 [CM503010]: Making route(s) to <sip:92211234@192.168.0.247>
    18:14:14.684 [MS210000] C:3.1:Offer received. RTP connection: 192.168.0.20:16462(16463)
    18:14:14.684 Remote SDP is set for legC:3.1
    18:14:14.684 [CM505001]: Ext.203: Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, no-replaces, able-no-sdp, recvonly] UserAgent: [Linksys/SPA3102-3.3.6(GW)] PBX contact: [sip:203@192.168.0.247:5060]
    18:14:14.653 [CM503001]: Call(3): Incoming call from Ext.203 to <sip:92211234@192.168.0.247>
    18:14:14.653 [CM500002]: Info on incoming INVITE:
    INVITE sip:92211234@192.168.0.247 SIP/2.0
    Via: SIP/2.0/UDP 192.168.0.20:5060;branch=z9hG4bK-9549277e
    Max-Forwards: 70
    Contact: "sales"<sip:203@192.168.0.20:5060>
    To: <sip:92211234@192.168.0.247>
    From: "sales"<sip:203@192.168.0.247>;tag=d503163d2951272ao0
    Call-ID: 271295f9-b8f34a9e@192.168.0.20
    CSeq: 102 INVITE
    Expires: 240
    Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
    Proxy-Authorization: Digest username="203",realm="3CXPhoneSystem",nonce="414d535c0167687638:3d92f701f509dcfbb99d0a9b08a29037",uri="sip:92211234@192.168.0.247",algorithm=MD5,response="5c24f91fd225577a334f3e49a9b69254"
    Supported: x-sipura
    User-Agent: Linksys/SPA3102-3.3.6(GW)
    Content-Length: 0
    Remote-Party-ID: sales <sip:203@192.168.0.247>;screen=yes;party=calling

    18:13:49.509 [CM503008]: Call(2): Call is terminated
    18:13:43.299 Session 59 of leg C:2.1 is confirmed
    18:13:43.189 [CM503007]: Call(2): Device joined: sip:EndCall@127.0.0.1:40600;rinstance=26223c814f7ae7d6
    18:13:43.189 [CM503007]: Call(2): Device joined: sip:203@192.168.0.20:5060
    18:13:43.174 [MS210005] C:2.1:Answer provided. Connection(proxy mode):192.168.0.247:7006(7007)
    18:13:43.174 [MS210001] C:2.3:Answer received. RTP connection[unsecure]: 127.0.0.1:40612(40613)
    18:13:43.174 Remote SDP is set for legC:2.3
    18:13:43.174 [CM503002]: Call(2): Alerting sip:EndCall@127.0.0.1:40600;rinstance=26223c814f7ae7d6
    18:13:43.033 [CM503025]: Call(2): Calling Unknown:Ext.EndCall@[Dev:sip:EndCall@127.0.0.1:40600;rinstance=26223c814f7ae7d6]
    18:13:43.033 [MS210004] C:2.3:Offer provided. Connection(proxy mode): 127.0.0.1:7010(7011)
    18:13:42.971 [CM503016]: Call(2): Attempt to reach <sip:2211234@192.168.0.247> failed. Reason: Server Failure
    18:13:42.955 [CM503003]: Call(2): Call to sip:2211234@192.168.0.20:5061 has failed; Cause: 503 Service Unavailable; from IP:192.168.0.20:5061
    18:13:42.893 [CM503025]: Call(2): Calling PSTNline:2211234@(Ln.10000@2213167)@[Dev:sip:10000@192.168.0.20:5061]
    18:13:42.877 [MS210006] C:2.2:Offer provided. Connection(by pass mode): 192.168.0.20:16460(16461)
    18:13:42.862 [CM503017]: Call(2): Target is not registered: PSTNline:2211234 dialed on (AnyLine@2210821)
    18:13:42.862 [CM303003]: There are no available outbound lines on gateway 2210821 at this time.
    18:13:42.862 [CM503004]: Call(2): Route 1: PSTNline:2211234@(Ln.10000@2213167)@[Dev:sip:10000@192.168.0.20:5061]
    18:13:42.799 [CM503010]: Making route(s) to <sip:2211234@192.168.0.247>
    18:13:42.799 [MS210000] C:2.1:Offer received. RTP connection: 192.168.0.20:16460(16461)
    18:13:42.799 Remote SDP is set for legC:2.1
    18:13:42.799 [CM505001]: Ext.203: Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, no-replaces, able-no-sdp, recvonly] UserAgent: [Linksys/SPA3102-3.3.6(GW)] PBX contact: [sip:203@192.168.0.247:5060]
    18:13:42.752 [CM503001]: Call(2): Incoming call from Ext.203 to <sip:2211234@192.168.0.247>
    18:13:42.721 [CM500002]: Info on incoming INVITE:
    INVITE sip:2211234@192.168.0.247 SIP/2.0
    Via: SIP/2.0/UDP 192.168.0.20:5060;branch=z9hG4bK-404ce055
    Max-Forwards: 70
    Contact: "sales"<sip:203@192.168.0.20:5060>
    To: <sip:2211234@192.168.0.247>
    From: "sales"<sip:203@192.168.0.247>;tag=2fb4338cb87e84e1o0
    Call-ID: 61fe7d28-45a167f5@192.168.0.20
    CSeq: 102 INVITE
    Expires: 240
    Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
    Proxy-Authorization: Digest username="203",realm="3CXPhoneSystem",nonce="414d535c0167685692:13c5ef690dc6e35a70ded3a88a5290cd",uri="sip:2211234@192.168.0.247",algorithm=MD5,response="7dc5cd4d49231ba383477e77b6b33bd4"
    Supported: x-sipura
    User-Agent: Linksys/SPA3102-3.3.6(GW)
    Content-Length: 0
    Remote-Party-ID: sales <sip:203@192.168.0.247>;screen=yes;party=calling

    18:10:54.115 [CM504001]: Ext.IVRForward: new contact is registered. Contact(s): [sip:IVRForward@127.0.0.1:40600;rinstance=67a2503d1b33236f/IVRForward]
    18:10:54.115 [CM504001]: Ext.800: new contact is registered. Contact(s): [sip:800@127.0.0.1:40600;rinstance=89cc8c9ff374de72/800]
    18:10:34.717 [CM504001]: Ext.206: new contact is registered. Contact(s): [sip:206@192.168.0.23:5061/206]
    18:10:33.936 [CM504001]: Ext.*1: new contact is registered. Contact(s): [sip:*1@127.0.0.1:40000;rinstance=2d5f7406eeb4b983/*1]
    18:10:31.842 [CM504001]: Ext.203: new contact is registered. Contact(s): [sip:203@192.168.0.20:5060/203]
    18:10:24.688 [EC200004]: IVR server is connected: application:win2k3x:0/IVRServer local:127.0.0.1:5482 remote:127.0.0.1:1035
    18:09:44.199 IP(s) added:[192.168.0.247]
    18:09:43.246 [CM306002]: There is no valid STUN server specified! External IP can not be resolved.
    18:09:40.996 [CM306001]: Address of STUN server (stun.3cx.com) could not be resolved!
    18:09:34.793 Failed to obtain short path name for [C:\ProgramData\3CX\Bin\Cert]
    18:09:34.152 [CM501002]: Version: 8.0.9318.0
    18:09:34.090 [CM501007]: *** Started Calls Controller thread ***
     
  2. rodster

    Joined:
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    I finally found the problem:
    -Apparently, the number of trunks (in the 3CX PSTN gateway settings) for the SPA3102 was set to more than 1, where SPA3102 has only one trunk. After setting the correct number of trunks for SPA 3102, the problem is gone.

    Although I do find this as a bug in 3CX. Though the number of trunks configured in 3CX per SPA3102 is incorrect, 3CX still knows that there are no more available trunk; this is evident as the extension who made an outside call that failed do get the message that the call cannot get through. However, the existing call that used the trunk hears the auto-attendant audio while the call is on-going and is very annoying.

    I can replicate the scenario by:
    -chaninging the number of trunks in 3CX PSTN gateway settings to any number greater than 1
    -while the trunk is being used, via an incoming call to an extension, another extension tries to make an outside call
    -the other extension gets the message that there are no more avaible trunks for the call
    -and when that other extension puts down the phone, the auto attendant audio interferes with the current call that used the trunk.

    My case closed but seems this bug should be fixed in the next release of 3CX.
     
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