RESOLVED ** Can't call 800/866/888 numbers...sometimes.

Discussion in '3CX Phone System - General' started by AdgTech, Dec 3, 2010.

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  1. AdgTech

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    I've got a pretty strange issue here. I've got a ticket open with Support, but I wanted to rack the brains of the community (and maybe leave a solution for someone in the future.)

    Sometimes I can call an 800 number and sometimes I can't. I can call the same number from two different phones and one will connect, the other will be "forbidden by administrator". Additionally, if the number is longer than 11 digits, it is "forbidden by administrator" (an example of this would be something like 1-888-VoIPSupply.)

    I've got a single outgoing rule that allows any call between 7 and 22 digits to be routed over a Patton Smartnode SN4960, configured by the 3CX software. I have not done anything outside of the stock configuration on the SmartNode.

    An example of what's going on is:

    I can call 1-888-427-7895, but my boss cannot. He can call 1-888-877-1655, but I cannot. Our fax machine using a HandyTone ATA can not call any of them. Whenever the call fails, wireshark shows a packet coming from the smartnode marked as "Rejected". Over the phone, I get the "Call forbidden by administrator." Unfortunately, I'm the administrator and I'm not forbidding anything. Everything else works on all of these phones. It's simply 800/866/888 numbers.

    I'm on V9 SP4 on Server 2008R2 (X64) fully patched and read to go. Unfortunately, I'm not sure if this worked prior to today or not as no one around here tells me something is broken until they needed it 5 minutes ago. :roll: If anyone has any ideas, I'm dying to hear them.
     
  2. leejor

    leejor Well-Known Member

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    Re: Can't call 800/866/888 numbers...sometimes. Call forbid

    Some 3CX logs showing the same number dialled from two different extensions, one failing and one working , might help us in analysing the problem. Offhand, I would suspect an outbound rule issue.
     
  3. AdgTech

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    Re: Can't call 800/866/888 numbers...sometimes. Call forbid

    Here's a quick log snippet of a normal formatted 888 number failing and succeeding. The top call succeeds, the bottom one fails.

    Code:
    21:56:33.503  [CM503008]: Call(2): Call is terminated
    21:56:28.841  [MS210003] C:2.2:Answer provided. Connection(transcoding mode[unsecure]):10.68.1.20:7008(7009)
    21:56:28.840  [MS210001] C:2.1:Answer received. RTP connection[unsecure]: 10.68.1.142:11800(11801)
    21:56:28.838  Remote SDP is set for legC:2.1
    21:56:28.121  [MS210002] C:2.1:Offer provided. Connection(transcoding mode): 10.68.1.20:7006(7007)
    21:56:28.112  Remote SDP is set for legC:2.2
    21:56:27.071  Session 954 of leg C:2.1 is confirmed
    21:56:26.889  [CM503007]: Call(2): Device joined: sip:10000@10.68.1.40:5060
    21:56:26.872  [CM503007]: Call(2): Device joined: sip:249@10.68.1.142:5062
    21:56:26.867  [MS210003] C:2.1:Answer provided. Connection(transcoding mode[unsecure]):10.68.1.20:7006(7007)
    21:56:26.859  [MS210001] C:2.2:Answer received. RTP connection[unsecure]: 10.68.1.40:5202(5203)
    21:56:26.857  Remote SDP is set for legC:2.2
    21:56:23.683  [CM503002]: Call(2): Alerting sip:10000@10.68.1.40:5060
    21:56:23.560  [CM503025]: Call(2): Calling Unknown:18884732963@(Ln.10000@SmartNode)@[Dev:sip:10000@10.68.1.40:5060]
    21:56:23.557  [MS210002] C:2.2:Offer provided. Connection(transcoding mode): 10.68.1.20:7008(7009)
    21:56:23.544  [CM503004]: Call(2): Route 1: Unknown:18884732963@(Ln.10000@SmartNode)@[Dev:sip:10000@10.68.1.40:5060]
    21:56:23.538  [CM503010]: Making route(s) to <sip:18884732963@10.68.1.20>
    21:56:23.538  [MS210000] C:2.1:Offer received. RTP connection: 10.68.1.142:11800(11801)
    21:56:23.535  Remote SDP is set for legC:2.1
    21:56:23.535  [CM505001]: Ext.249: Device info: Device Identified: [Man: Yealink;Mod: T28;Rev: General] Capabilities:[reinvite, no-replaces, unable-no-sdp, no-recvonly] UserAgent: [Yealink SIP-T28P 2.51.0.10] PBX contact: [sip:249@10.68.1.20:5060]
    21:56:23.505  [CM503001]: Call(2): Incoming call from Ext.249 to <sip:18884732963@10.68.1.20>
    21:56:23.492  [CM500002]: Info on incoming INVITE:
      INVITE sip:18884732963@10.68.1.20 SIP/2.0
      Via: SIP/2.0/UDP 10.68.1.142:5062;rport=5062;branch=z9hG4bK1848626162
      Max-Forwards: 70
      Contact: <sip:249@10.68.1.142:5062>
      To: <sip:18884732963@10.68.1.20>
      From: "Brad "<sip:249@10.68.1.20>;tag=771807564
      Call-ID: 1757448185@10.68.1.142
      CSeq: 2 INVITE
      Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
      Proxy-Authorization: Digest username="Brad",realm="3CXPhoneSystem",nonce="414d535c030a4ce720:49a9f9e6c2f39ff0093ff4abafde2f0c",uri="sip:18884732963@10.68.1.20",response="e89fc376c850f911038bb14e48294b47",algorithm=MD5
      Supported: replaces
      User-Agent: Yealink SIP-T28P 2.51.0.10
      Allow-Events: talk, hold, conference, refer, check-sync
      Content-Length: 0
      
    21:56:15.260  Currently active calls [none]
    21:55:43.266  Currently active calls [none]
    21:55:28.826  [CM503008]: Call(1): Call is terminated
    21:55:15.437  Session 933 of leg C:1.1 is confirmed
    21:55:15.343  [CM503007]: Call(1): Device joined: sip:EndCall@127.0.0.1:40600;rinstance=f36e49bc85107c17
    21:55:15.326  [CM503007]: Call(1): Device joined: sip:887@10.68.1.132;user=phone
    21:55:15.324  [MS210007] C:1.1:Answer provided. Connection(by pass mode): 10.68.1.20:40610(40611)
    21:55:15.317  [MS210001] C:1.3:Answer received. RTP connection[unsecure]: 10.68.1.20:40610(40611)
    21:55:15.315  Remote SDP is set for legC:1.3
    21:55:15.312  [CM503002]: Call(1): Alerting sip:EndCall@127.0.0.1:40600;rinstance=f36e49bc85107c17
    21:55:13.931  [CM503025]: Call(1): Calling Unknown:Ext.EndCall@[Dev:sip:EndCall@127.0.0.1:40600;rinstance=f36e49bc85107c17]
    21:55:13.928  [MS210006] C:1.3:Offer provided. Connection(by pass mode): 10.68.1.132:5004(5005)
    21:55:13.864  [CM503016]: Call(1): Attempt to reach <sip:18884732963@10.68.1.20;user=phone> failed. Reason: Forbidden
    21:55:13.795  [CM503003]: Call(1): Call to sip:18884732963@10.68.1.40:5060 has failed; Cause: 403 Forbidden; from IP:10.68.1.40:5060
    21:55:12.459  [CM503002]: Call(1): Alerting sip:10000@10.68.1.40:5060
    21:55:12.308  [CM503025]: Call(1): Calling Unknown:18884732963@(Ln.10000@SmartNode)@[Dev:sip:10000@10.68.1.40:5060]
    21:55:12.297  [MS210006] C:1.2:Offer provided. Connection(by pass mode): 10.68.1.132:5004(5005)
    21:55:12.275  [CM503004]: Call(1): Route 1: Unknown:18884732963@(Ln.10000@SmartNode)@[Dev:sip:10000@10.68.1.40:5060]
    21:55:12.244  [MS210000] C:1.1:Offer received. RTP connection: 10.68.1.132:5004(5005)
    21:55:12.234  [CM503010]: Making route(s) to <sip:18884732963@10.68.1.20;user=phone>
    21:55:12.229  Remote SDP is set for legC:1.1
    21:55:12.218  [CM503001]: Call(1): Incoming call from Fax.887 to <sip:18884732963@10.68.1.20;user=phone>
    21:55:12.045  [CM500002]: Info on incoming INVITE:
      INVITE sip:18884732963@10.68.1.20;user=phone SIP/2.0
      Via: SIP/2.0/UDP 10.68.1.132;branch=z9hG4bK6c0544b932955ee4
      Max-Forwards: 70
      Contact: <sip:887@10.68.1.132;user=phone>
      To: <sip:18884732963@10.68.1.20;user=phone>
      From: "Standard Fax Machine"<sip:887@10.68.1.20;user=phone>;tag=a4fb95db16fb95db
      Call-ID: 7f78e027feb3ccf7@10.68.1.132
      CSeq: 15334 INVITE
      Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, INFO, SUBSCRIBE, UPDATE
      Proxy-Authorization: Digest username="887",realm="3CXPhoneSystem",algorithm=MD5,uri="sip:18884732963@10.68.1.20;user=phone",nonce="414d535c030a4c9f05:d20953d56cf86fa5d8ae7dd8d5ae1c17",response="3205a6bdf2756ca01f625f3a5bca5ffa"
      Supported: replaces, timer
      User-Agent: Grandstream HT287 1.1.0.37
      Content-Length: 0
    
    As for the outbound rule, I have one. It's anything between 7 and 22 digits goes out to the smartnode. That's it, there's no prepending/stripping/nothing. It's just that.

    Thanks for the reply!
     
  4. SY

    SY Well-Known Member
    3CX Support

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    Re: Can't call 800/866/888 numbers...sometimes. Call forbid

    21:55:13.795 [CM503003]: Call(1): Call to sip:18884732963@10.68.1.40:5060 has failed; Cause: 403 Forbidden; from IP:10.68.1.40:5060

    You need to check Patton configuration. Also Patton logs should specify reason why call is forbidden by the gateway.

    Thanks
     
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  5. leejor

    leejor Well-Known Member

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    Re: Can't call 800/866/888 numbers...sometimes. Call forbid

    Besides checking the Patton logs for the forbidden reason, you may want to re-think your outbound rules for North America, making them a bit less "broad" If someone is meant to dial a 1+ 10 digit number something should be truncating the number if they add on extra digits. If you ever introduce VoIP trunks at a later date it could become a problem. It might be a problem now with the Patton as it may have an internal dial plan. Many ATA's and VoIP phones allow you to set up a device dialplan to "groom" numbers to the correct length before being sent to the PBX (or provider).
     
  6. AdgTech

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    Once again the amazing 3cx support team comes through (thanks William!).

    The outbound caller ID for the actual smartnode was not set. (PSTN Devices -> Device -> Virtual Extension). Setting the CID made everything work. I just have no idea why.

    Leejor:

    The rule I have right now is just a "let's get this thing running" rule. I'm actually going to be adding some VoIP trunks in the near future, so paring down the rules is next on my task list.

    How would I go about setting a truncate rule? It doesn't really seem possible in 3CX. Would I do that on the patton?
     
  7. leejor

    leejor Well-Known Member

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    In many cases can use the internal dial plan within the VoIP phones or ATA's that you are using. Dial plans have to accommodate all digit patterns that a user may dial. Most people use them because they eliminate the 4 to 5 second wait after dialling the last digit, or having to hit # art the end to "speed the call along". One part of the dial plan (for NA LD calls) could include a rule so that a if a caller dialled 91XXXxxxXXXX the set would recognise that no other digits should be added to that and send it to 3CX. Depending on the make of your sets the dial plan can get quite creative (complicated), so it is good idea to really plan this out and implement it (test) on one phone before rolling out to the whole office.
     
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