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Call transfer failed, number is busy

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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:[email protected]: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:[email protected]: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:[email protected]:5060]
14:11:28.095  Refer: from=<sip:[email protected]:5060>;tag=b45b295b; to="8888888888"<sip:[email protected]:5060>;tag=8d65b33a; RefTo=<sip:[email protected]: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:[email protected]:40600;rinstance=748fb13dbed87242
14:11:17.093  [CM503007]: Call(59): Device joined: sip:[email protected]: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:[email protected]:5060]
14:11:17.086  [CM503002]: Call(59): Alerting sip:[email protected]:40600;rinstance=748fb13dbed87242
14:11:16.935  [CM503025]: Call(59): Calling Ext:Ext.1080@[Dev:sip:[email protected]: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:[email protected]:40600;rinstance=748fb13dbed87242]
14:11:16.898  [CM503010]: Making route(s) to <sip:[email protected]: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:[email protected]:5060]
14:11:16.889  [CM503001]: Call(59): Incoming call from 4502880515@(Ln.10002@Ubity) to <sip:[email protected]: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:[email protected]:5060 SIP/2.0
  Via: SIP/2.0/UDP 0.1.2.3.4:5060;branch=z9hG4bK267ef389;rport=5060
  Max-Forwards: 70
  Contact: <sip:[email protected]>
  To: <sip:[email protected]:5060>
  From: "8888888888"<sip:[email protected]>;tag=as1d4d48ab
  Call-ID: [email protected]
  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
 
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:[email protected]:5060]
13:40:50.546  [CM503002]: Call(17): Alerting sip:[email protected]:5060
13:40:50.046  [CM503025]: Call(17): Calling PSTNline:06102913xx@(Ln.12002@GSM Gateway)@[Dev:sip:[email protected]:5060]
13:40:49.984  [CM503004]: Call(17): Route 1: PSTNline:06102913xx@(Ln.12002@GSM Gateway)@[Dev:sip:[email protected]:5060]
13:40:49.984  [CM503010]: Making route(s) to <sip:[email protected]:5060>
13:40:37.171  Currently active calls - 1: [17]
13:40:36.718  [CM503007]: Call(17): Device joined: sip:[email protected]:40600;rinstance=5744e94622ba8068
13:40:36.718  [CM503007]: Call(17): Device joined: sip:[email protected]: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:[email protected]:5060]
13:40:36.718  [CM503002]: Call(17): Alerting sip:[email protected]:40600;rinstance=5744e94622ba8068
13:40:36.562  [CM503025]: Call(17): Calling Ext:Ext.802@[Dev:sip:[email protected]:40600;rinstance=5744e94622ba8068]
13:40:36.515  [CM503004]: Call(17): Route 1: Ext:Ext.802@[Dev:sip:[email protected]:40600;rinstance=5744e94622ba8068]
13:40:36.515  [CM503010]: Making route(s) to <sip:[email protected]>
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:[email protected]:5060]
13:40:36.515  [CM503001]: Call(17): Incoming call from Ext.101 to <sip:[email protected]>

13:40:18.656  [CM503008]: Call(16): Call is terminated
13:40:13.140  [CM503016]: Call(16): Attempt to reach <sip:[email protected]: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:[email protected]: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:[email protected]: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:[email protected]:40600;rinstance=5744e94622ba8068
13:39:59.031  [CM503007]: Call(16): Device joined: sip:[email protected]: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:[email protected]:5060]
13:39:59.031  [CM503002]: Call(16): Alerting sip:[email protected]:40600;rinstance=5744e94622ba8068
13:39:58.906  [CM503025]: Call(16): Calling Ext:Ext.802@[Dev:sip:[email protected]:40600;rinstance=5744e94622ba8068]
13:39:58.843  [CM503004]: Call(16): Route 1: Ext:Ext.802@[Dev:sip:[email protected]:40600;rinstance=5744e94622ba8068]
13:39:58.843  [CM503010]: Making route(s) to <sip:[email protected]: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:[email protected]:5060]
13:39:58.828  [CM503001]: Call(16): Incoming call from 06102913xx@(Ln.10002@CCS) to <sip:[email protected]:5060>
13:39:58.593  [CM503012]: Inbound out-of-office hours rule (unnamed) for 10002 forwards to DN:802
 
Anyone ?

I'm sure it's just a little detail I missed...
 
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.
 
LeonidasG said:
Go to your Outbound Rule for the gateway / provider you are using and remove any Extension Restrictions you have added.
Click to expand...

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.
 
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
 
Hi guys,

LeonidasG, many thanks for your answer.

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

LeonidasG said:
I'm guessing your outbound rule for that gateway has some restrictions restricting specific extensions from making call.
Am i correct?
Click to expand...
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!
 
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 !
 
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:[email protected]: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:[email protected]:5060> failed. Reason: Busy
15:56:50.984 [CM503003]: Call(2): Call to sip:[email protected]: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:[email protected]:5060]
15:56:50.671 [CM503005]: Call(2): Forwarding: VoIPline:1416XXXXXXX@(Ln.10001@FPL)@[Dev:sip:[email protected]: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:[email protected]:5060]
15:56:47.406 [CM503002]: Call(2): Alerting sip:[email protected]: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:[email protected]:2250;rinstance=92e2dc84e05a27b3]
15:56:45.578 [CM503004]: Call(2): Route 1: Ext:Ext.100@[Dev:sip:[email protected]:2250;rinstance=92e2dc84e05a27b3]
15:56:45.562 [CM503010]: Making route(s) to <sip:[email protected]: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:[email protected]: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:[email protected]:40600;rinstance=5ffa1e04f7fbc3dd
15:56:31.562 [CM503007]: Call(2): Device joined: sip:[email protected]: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:[email protected]:5060]
15:56:31.546 [CM503002]: Call(2): Alerting sip:[email protected]:40600;rinstance=5ffa1e04f7fbc3dd
15:56:31.343 [CM503025]: Call(2): Calling Ext:Ext.800@[Dev:sip:[email protected]:40600;rinstance=5ffa1e04f7fbc3dd]
15:56:31.312 [CM503004]: Call(2): Route 1: Ext:Ext.800@[Dev:sip:[email protected]:40600;rinstance=5ffa1e04f7fbc3dd]
15:56:31.312 [CM503010]: Making route(s) to <sip:[email protected]: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:[email protected]:52268]
15:56:31.281 [CM503001]: Call(2): Incoming call from 647XXXXXXX@(Ln.10001@FPL) to <sip:[email protected]:5060>
15:56:31.234 [CM503012]: Inbound out-of-office hours rule (unnamed) for 10001 forwards to DN:800
 
Is the call coming in on the same VOIP line as you are trying to dial out on?
 
yes
 
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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