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Call transfer failed, number is busy

Discussion in '3CX Phone System - General' started by chrischevy, Sep 24, 2010.

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  1. chrischevy

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    Hi,

    When the out of office rule is on, incoming calls are not redirected to mobile number. The call gets the "Call transfer Failed, number is busy" message. Calling the same number from an internal phone works fine.
    Like many others on this forum, I see "There's no outbound rule for external number" in the log file.
    I defined a general outbound rule with no extension, no number length and no prefix: it still doesn't work...
    I even tried calling inbound on a different VOIP provider.

    Any idea ?

    (I changed sensitive information in the log)

    Code:
    14:11:33.159  [CM503008]: Call(59): Call is terminated
    14:11:28.249  [MS210003] C:59.2:Answer provided. Connection(transcoding mode[unsecure]):127.0.0.1:7158(7159)
    14:11:28.248  [MS210000] C:59.2:Offer received. RTP connection: 127.0.0.1:40650(40651)
    14:11:28.246  Remote SDP is set for legC:59.2
    14:11:28.131  [CM503016]: Call(59): Attempt to reach <sip:1085@127.0.0.1:5060> failed. Reason: Busy
    14:11:28.130  [CM303006]: There's no outbound rule for external number '9999999999' defined
    14:11:28.114  [CM303006]: There's no outbound rule for external number '9999999999' defined
    14:11:28.097  [CM503010]: Making route(s) to <sip:1085@127.0.0.1:5060>
    14:11:28.095  [CM505003]: Provider:[XXXXXX] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [Asterisk PBX] PBX contact: [sip:xxxxxxxx@0.1.2.3:5060]
    14:11:28.095  Refer: from=<sip:1080@127.0.0.1:5060>;tag=b45b295b; to="8888888888"<sip:8888888888@127.0.0.1:5060>;tag=8d65b33a; RefTo=<sip:1085@127.0.0.1:5060>
    14:11:27.983  [MS210003] C:59.2:Answer provided. Connection(transcoding mode[unsecure]):127.0.0.1:7158(7159)
    14:11:27.982  [MS210000] C:59.2:Offer received. RTP connection: 127.0.0.1:40650(40651)
    14:11:27.980  Remote SDP is set for legC:59.2
    14:11:22.151  [MS211000] C:59.1: 0.1.2.3:17976 is delivering DTMF using RTP payload (RFC2833). In-Band DTMF tone detection is disabled for this call segment.
    14:11:17.193  Session 16417 of leg C:59.1 is confirmed
    14:11:17.098  [CM503007]: Call(59): Device joined: sip:1080@127.0.0.1:40600;rinstance=748fb13dbed87242
    14:11:17.093  [CM503007]: Call(59): Device joined: sip:xxxxxxxx@xxxxxxx.com:5060
    14:11:17.091  [MS210003] C:59.1:Answer provided. Connection(transcoding mode[unsecure]):68.67.40.125:9030(9031)
    14:11:17.090  [MS210001] C:59.2:Answer received. RTP connection[unsecure]: 127.0.0.1:40650(40651)
    14:11:17.088  Remote SDP is set for legC:59.2
    14:11:17.086  [CM505001]: Ext.1080: Device info: Device Identified: [Man: 3CX Ltd.;Mod: 3CX IVR;Rev: General] Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [3CX IVR] PBX contact: [sip:1080@127.0.0.1:5060]
    14:11:17.086  [CM503002]: Call(59): Alerting sip:1080@127.0.0.1:40600;rinstance=748fb13dbed87242
    14:11:16.935  [CM503025]: Call(59): Calling Ext:Ext.1080@[Dev:sip:1080@127.0.0.1:40600;rinstance=748fb13dbed87242]
    14:11:16.934  [MS210002] C:59.2:Offer provided. Connection(transcoding mode): 127.0.0.1:7158(7159)
    14:11:16.899  [CM503004]: Call(59): Route 1: Ext:Ext.1080@[Dev:sip:1080@127.0.0.1:40600;rinstance=748fb13dbed87242]
    14:11:16.898  [CM503010]: Making route(s) to <sip:1080@10.0.0.66:5060>
    14:11:16.898  [MS210000] C:59.1:Offer received. RTP connection: 0.1.2.3:17976(17977)
    14:11:16.897  Remote SDP is set for legC:59.1
    14:11:16.897  [CM505003]: Provider:[XXXXX] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [Asterisk PBX] PBX contact: [sip:xxxxxxxx@0.1.2.3:5060]
    14:11:16.889  [CM503001]: Call(59): Incoming call from 4502880515@(Ln.10002@Ubity) to <sip:1080@10.0.0.66:5060>
    14:11:16.878  [CM503012]: Inbound office hours rule (unnamed) for 10002 forwards to DN:1080
    14:11:16.878  Looking for inbound target: called=4503226259; caller=4502880515
    14:11:16.874  [CM500002]: Info on incoming INVITE:
      INVITE sip:8888888888@0.1.2.3:5060 SIP/2.0
      Via: SIP/2.0/UDP 0.1.2.3.4:5060;branch=z9hG4bK267ef389;rport=5060
      Max-Forwards: 70
      Contact: <sip:8888888888@0.1.2.3>
      To: <sip:8888888888@0.1.2.3:5060>
      From: "8888888888"<sip:8888888888@0.1.2.3>;tag=as1d4d48ab
      Call-ID: 29f41fe7734fadff7bd2ae70113ace5e@0.1.2.3
      CSeq: 102 INVITE
      Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
      Date: Fri, 24 Sep 2010 18:11:14 GMT
      Supported: replaces
      User-Agent: Asterisk PBX
      Content-Length: 0
    
     
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  2. complex1

    complex1 Active Member

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    Hi,

    I have the same problem over here.

    I have setup a DR with the virtual extension 802
    In the DR, Ext. 201 is selected.

    Setup Ext. 201
    Under Forwarding Rules > Out of Office > Forward Internal and External Calls > select “An external number” with 06102913xx.
    Under Other > User Information > Current status > select Out of Office

    When dialling the DR from internal extension 101, the call transferred well, but when the call is from the VoIP provider into DR the call failed and number is busy.
    For the outgoing calls I use a Portech GSM gateway.

    Code:
    13:41:00.250  [CM503008]: Call(17): Call is terminated
    13:40:50.546  [CM505002]: Gateway:[GSM Gateway] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [CM5K  (706220)] PBX contact: [sip:12002@172.17.84.34:5060]
    13:40:50.546  [CM503002]: Call(17): Alerting sip:12002@172.17.84.251:5060
    13:40:50.046  [CM503025]: Call(17): Calling PSTNline:06102913xx@(Ln.12002@GSM Gateway)@[Dev:sip:12002@172.17.84.251:5060]
    13:40:49.984  [CM503004]: Call(17): Route 1: PSTNline:06102913xx@(Ln.12002@GSM Gateway)@[Dev:sip:12002@172.17.84.251:5060]
    13:40:49.984  [CM503010]: Making route(s) to <sip:201@127.0.0.1:5060>
    13:40:37.171  Currently active calls - 1: [17]
    13:40:36.718  [CM503007]: Call(17): Device joined: sip:802@127.0.0.1:40600;rinstance=5744e94622ba8068
    13:40:36.718  [CM503007]: Call(17): Device joined: sip:101@172.17.84.253:5060;line=63816
    13:40:36.718  [CM505001]: Ext.802: Device info: Device Identified: [Man: 3CX Ltd.;Mod: 3CX IVR;Rev: General] Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [3CX IVR] PBX contact: [sip:802@127.0.0.1:5060]
    13:40:36.718  [CM503002]: Call(17): Alerting sip:802@127.0.0.1:40600;rinstance=5744e94622ba8068
    13:40:36.562  [CM503025]: Call(17): Calling Ext:Ext.802@[Dev:sip:802@127.0.0.1:40600;rinstance=5744e94622ba8068]
    13:40:36.515  [CM503004]: Call(17): Route 1: Ext:Ext.802@[Dev:sip:802@127.0.0.1:40600;rinstance=5744e94622ba8068]
    13:40:36.515  [CM503010]: Making route(s) to <sip:802@172.17.84.34>
    13:40:36.515  [CM505001]: Ext.101: Device info: Device Identified: [Man: Snom;Mod: M3;Rev: General] Capabilities:[reinvite, replaces, unable-no-sdp, no-recvonly] UserAgent: [snom-m3-SIP/02.11 (MAC=0004132A32FB; HW=1)] PBX contact: [sip:101@172.17.84.34:5060]
    13:40:36.515  [CM503001]: Call(17): Incoming call from Ext.101 to <sip:802@172.17.84.34>
    
    13:40:18.656  [CM503008]: Call(16): Call is terminated
    13:40:13.140  [CM503016]: Call(16): Attempt to reach <sip:201@127.0.0.1:5060> failed. Reason: Busy
    13:40:13.140  [CM303006]: There's no outbound rule for external number '06533403xx' defined
    13:40:13.140  [CM303006]: There's no outbound rule for external number '06533403xx' defined
    13:40:13.140  [CM503010]: Making route(s) to <sip:201@127.0.0.1:5060>
    13:40:13.140  [CM505003]: Provider:[CCS] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [GW01.budgetphone.nl] PBX contact: [sip:2236688xx@84.105.39.42:5060]
    13:40:07.390  [MS211000] C:16.1: 83.143.188.165:58336 is delivering DTMF using RTP payload (RFC2833). In-Band DTMF tone detection is disabled for this call segment.
    13:40:05.171  Currently active calls - 1: [16]
    13:39:59.031  [CM503007]: Call(16): Device joined: sip:802@127.0.0.1:40600;rinstance=5744e94622ba8068
    13:39:59.031  [CM503007]: Call(16): Device joined: sip:2236688xx@sip1.budgetphone.nl:5060
    13:39:59.031  [CM505001]: Ext.802: Device info: Device Identified: [Man: 3CX Ltd.;Mod: 3CX IVR;Rev: General] Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [3CX IVR] PBX contact: [sip:802@127.0.0.1:5060]
    13:39:59.031  [CM503002]: Call(16): Alerting sip:802@127.0.0.1:40600;rinstance=5744e94622ba8068
    13:39:58.906  [CM503025]: Call(16): Calling Ext:Ext.802@[Dev:sip:802@127.0.0.1:40600;rinstance=5744e94622ba8068]
    13:39:58.843  [CM503004]: Call(16): Route 1: Ext:Ext.802@[Dev:sip:802@127.0.0.1:40600;rinstance=5744e94622ba8068]
    13:39:58.843  [CM503010]: Making route(s) to <sip:802@172.17.84.34:5060>
    13:39:58.843  [CM505003]: Provider:[CCS] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [GW01.budgetphone.nl] PBX contact: [sip:2236688xx@84.105.39.42:5060]
    13:39:58.828  [CM503001]: Call(16): Incoming call from 06102913xx@(Ln.10002@CCS) to <sip:802@172.17.84.34:5060>
    13:39:58.593  [CM503012]: Inbound out-of-office hours rule (unnamed) for 10002 forwards to DN:802
     
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  3. chrischevy

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    Anyone ?

    I'm sure it's just a little detail I missed...
     
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  4. LeonidasG

    LeonidasG Support Team
    Staff Member 3CX Support

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    Hi,

    I'm guessing your outbound rule for that gateway has some restrictions restricting specific extensions from making call.
    Am i correct?

    If this is the case, then you have unfortunately stumped on a bug in the phonesystem. We are planning to release a fix / Service Pack fixing this in the near future.
    But for now if i'm correct and we are talking about the same issue i think we are, there is 1 workaround you can do for now.

    Go to your Outbound Rule for the gateway / provider you are using and remove any Extension Restrictions you have added.
     
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  5. chrischevy

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    Thank you for your help LeonidasG.

    I already have a rule that matches any extension, with no prefix and no number length. I placed it last so it can catch all the calls that do not match the other rules. It doesn't work.
     
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  6. LeonidasG

    LeonidasG Support Team
    Staff Member 3CX Support

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    Sorry, our friend Complex1 hijacked the thread for a moment there and accidentally replied to him instead of you :mrgreen:

    In regards to your own issue, i'd suggest you check how many digits your outbound rule is striping and whether you have added an outbound prefix in the number you added in the forwarding rules for 1080
     
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  7. complex1

    complex1 Active Member

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    Hi guys,

    LeonidasG, many thanks for your answer.

    I am very sorry for hijacking this thread, my apologies for that.

    You are right; I have restrictions on this outbound rule.
    Ext 100-199,201 are allowed to use this rule
    After removing the Extensions all works fine now, so I think I stumped on a bug.
    I will look forward for the oncoming fix / SP.

    Kind regards!
     
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  8. chrischevy

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    I just got the answer from Arthur at 3CX.

    This bug will be effectively fixed on next update.

    Thank you all at 3CX for the great support !
     
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  9. kolagola

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    Hi,

    I have this same problem, transfer failed error 486, line busy.

    my scenario is that I have two extensions 100 & 101

    I am using 3cx softphone as ext 100 on desktop (3cx server is also installed on this machine) & eyebeam as ext 101 on laptop.

    Inbound is set to go directly in IVR, option 1 transfers call to ext 100(which works excellent)

    If unanswered within 10 sec, call to be forwarded to mobile that is 1416xxxxxxx, i get message transfer failed error 486, line busy.

    In the out bound rule for mobile,
    1.calls to nos. starting with perfix is set to 1
    2. calls from extension is left blank
    3. calls with nos with length left blank
    4. route is set to voip provider
    5. strip digits set to 0 since the voip provider requires the no. to be dialed with 1 prefix

    I am attaching the log for your detailed diagnosis. Please help..........

    15:56:59.687 [CM503008]: Call(2): Call is terminated
    15:56:51.593 [CM503003]: Call(2): Call to sip:100@192.168.1.102:5060 has failed; Cause: 487 Request Terminated; from IP:127.0.0.1:2250
    15:56:50.984 [CM503016]: Call(2): Attempt to reach <sip:100@127.0.0.1:5060> failed. Reason: Busy
    15:56:50.984 [CM503003]: Call(2): Call to sip:1416XXXXXXX@vbuzzer.com:5060 has failed; Cause: 486 1 Busy Here!; from IP:69.172.204.133:5060
    15:56:50.796 [CM503025]: Call(2): Calling VoIPline:1416XXXXXXX@(Ln.10001@FPL)@[Dev:sip:abcd@vbuzzer.com:5060]
    15:56:50.671 [CM503005]: Call(2): Forwarding: VoIPline:1416XXXXXXX@(Ln.10001@FPL)@[Dev:sip:abcd@vbuzzer.com:5060]
    15:56:47.406 [CM505001]: Ext.100: Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [3CXPhone 4.0.13679.0] PBX contact: [sip:100@127.0.0.1:5060]
    15:56:47.406 [CM503002]: Call(2): Alerting sip:100@127.0.0.1:2250;rinstance=92e2dc84e05a27b3
    15:56:45.906 Currently active calls - 1: [2]
    15:56:45.625 [CM503025]: Call(2): Calling Ext:Ext.100@[Dev:sip:100@127.0.0.1:2250;rinstance=92e2dc84e05a27b3]
    15:56:45.578 [CM503004]: Call(2): Route 1: Ext:Ext.100@[Dev:sip:100@127.0.0.1:2250;rinstance=92e2dc84e05a27b3]
    15:56:45.562 [CM503010]: Making route(s) to <sip:100@127.0.0.1:5060>
    15:56:45.562 [CM505003]: Provider:[FPL] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [vBuzzer UA] PBX contact: [sip:abcd@173.32.114.80:52268]
    15:56:40.562 [MS211000] C:2.1: 65.39.128.70:42302 is delivering DTMF using RTP payload (RFC2833). In-Band DTMF tone detection is disabled for this call segment.
    15:56:31.578 [CM503007]: Call(2): Device joined: sip:800@127.0.0.1:40600;rinstance=5ffa1e04f7fbc3dd
    15:56:31.562 [CM503007]: Call(2): Device joined: sip:abcd@vbuzzer.com:5060
    15:56:31.546 [CM505001]: Ext.800: Device info: Device Identified: [Man: 3CX Ltd.;Mod: 3CX IVR;Rev: General] Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [3CX IVR] PBX contact: [sip:800@127.0.0.1:5060]
    15:56:31.546 [CM503002]: Call(2): Alerting sip:800@127.0.0.1:40600;rinstance=5ffa1e04f7fbc3dd
    15:56:31.343 [CM503025]: Call(2): Calling Ext:Ext.800@[Dev:sip:800@127.0.0.1:40600;rinstance=5ffa1e04f7fbc3dd]
    15:56:31.312 [CM503004]: Call(2): Route 1: Ext:Ext.800@[Dev:sip:800@127.0.0.1:40600;rinstance=5ffa1e04f7fbc3dd]
    15:56:31.312 [CM503010]: Making route(s) to <sip:800@192.168.1.102:5060>
    15:56:31.312 [CM505003]: Provider:[FPL] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [vBuzzer UA] PBX contact: [sip:abcd@173.32.114.80:52268]
    15:56:31.281 [CM503001]: Call(2): Incoming call from 647XXXXXXX@(Ln.10001@FPL) to <sip:800@192.168.1.102:5060>
    15:56:31.234 [CM503012]: Inbound out-of-office hours rule (unnamed) for 10001 forwards to DN:800
     
  10. TwigsUSAN

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    Is the call coming in on the same VOIP line as you are trying to dial out on?
     
  11. kolagola

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  12. TwigsUSAN

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    That could be your problem. Does your VOIP provider allow multiple calls? If not, then you will never be able to do the transfer. You would need another VOIP line.
     
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