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Cannot transfer using Polycom phones?

Discussion in '3CX Phone System - General' started by Denny, Jul 6, 2007.

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  1. Denny

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    I have tried using Polycom IP 501 SIP and 330 SIP phones to transfer when I receive calls. I can call the extensions but I can't transfer.

    Here are the log:
    17:46:22.078 StratLink::eek:nHangUp [CM104001] Call(125): Ln:10000@Viatalk hung up call; cause: BYE; from IP:216.218.201.134
    17:46:18.500 CallConf::eek:nIncoming [CM003002] Call(126): Destination not available for call from sip:104@192.168.200.15 to sip:10@192.168.200.15
    17:46:18.500 CallConf::Rejected [CM103005] Call(126) is rejected: Destination does not exist, or is not registered
    17:46:18.484 CallConf::eek:nIncoming [CM103002] Call(126): Incoming call from 104 (Ext.104) to sip:10@192.168.200.15
    17:46:11.125 MediaServerReporting::RTPSender [MS004000] Call(125) Ext.104: (ACTIVE): Can't send RTP stream to 0.0.0.0:2228 destination unreachable
    17:46:07.687 CallLegImpl::eek:nConnected [CM103001] Call(125): Created audio channel for Ext.104 (192.168.200.113:2228) with Media Server (192.168.200.15:7330)
    17:46:07.671 StratInOut::eek:nConnected [CM104005] Call(125): Setup completed for call from Ln:10000@Viatalk to Ext.104
    17:46:07.656 CallLegImpl::eek:nConnected [CM103001] Call(125): Created audio channel for Ln:10000@Viatalk (216.218.201.134:17500) with Media Server (75.22.169.234:9000)
    17:46:03.515 CallConf::eek:nProvisional [CM103003] Call(125): Ext.104 is ringing
    17:46:03.203 CallConf::eek:nIncoming [CM103002] Call(125): Incoming call from 16198088696 (Ln:10000@Viatalk) to sip:16195797351@sanfrancisco-1.vtnoc.net:5060

    Our extensions are from 100-110. You can see Phone was picked up on 104 and tried transfering to 103. I see the error 17:46:18.484 CallConf::eek:nIncoming [CM103002] Call(126): Incoming call from 104 (Ext.104) to sip:10@192.168.200.15. Why is it going to sip:10@192.168.200.15?

    Any ideas?
     
  2. Denny

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    3CX Phone System Version3.0.2295.0
     
  3. DaKhalli

    DaKhalli New Member

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    try playing with the settings : supports reinvite and Support "Replace headers".

    Might solve the transfer issue.
     
  4. Denny

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    Under the VOIP Gateway, I enabled the "Specify how this voip provider handles transfers.

    Supports Re-Invite
    Supports 'Replaces' header

    Logs:

    19:23:30.328 StratLink::eek:nHangUp [CM104001] Call(12): Ln:10000@Viatalk hung up call; cause: BYE; from IP:216.218.201.134
    19:23:24.671 MediaServerReporting::RTPSender [MS104000] Call(12) Ext.105: (ACTIVE) RTP stream to 192.168.200.112:2222 restored.
    19:23:19.796 CallConf::eek:nIncoming [CM003002] Call(14): Destination not available for call from sip:105@192.168.200.15 to sip:10@192.168.200.15
    19:23:19.796 CallConf::Rejected [CM103005] Call(14) is rejected: Destination does not exist, or is not registered
    19:23:19.781 CallConf::eek:nIncoming [CM103002] Call(14): Incoming call from 105 (Ext.105) to sip:10@192.168.200.15
    19:23:17.640 MediaServerReporting::RTPSender [MS004000] Call(12) Ext.105: (ACTIVE): Can't send RTP stream to 0.0.0.0:2222 destination unreachable
    19:23:12.156 CallLegImpl::eek:nConnected [CM103001] Call(12): Created audio channel for Ext.105 (192.168.200.112:2222) with Media Server (192.168.200.15:7036)
    19:23:12.156 StratInOut::eek:nConnected [CM104005] Call(12): Setup completed for call from Ln:10000@Viatalk to Ext.105
    19:23:12.140 CallLegImpl::eek:nConnected [CM103001] Call(12): Created audio channel for Ln:10000@Viatalk (216.218.201.134:13720) with Media Server (192.168.200.15:9002)


    This doesn't effect incoming calls. It also effects internal extension transfers. What i mean is if I connect from one extention to another and try to transfer that call to another it also doesn't work.
     
  5. DaKhalli

    DaKhalli New Member

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    try setting it in your extensions properties
     
  6. Denny

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    It is enabled. I have tried it disabled also. Didn't work. For some reason when i put call on to transfer, when i try to input the extension, it dies after 2 digits. It beeps out after 2 digits and I can't finish. If I put call on hold, then hit new call it will allow me to call another extension but I can't conference at all.
     
  7. DaKhalli

    DaKhalli New Member

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    btw just for the record, you tryed disabling the supports reinvite in the providor settings too? else try it.
     
  8. Denny

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    I have it all disabled.

    19:46:11.585 StratLink::eek:nHangUp [CM104001] Call(21): Ln:10000@Viatalk hung up call; cause: BYE; from IP:216.218.201.134
    19:46:04.866 CallConf::eek:nIncoming [CM003002] Call(22): Destination not available for call from sip:105@192.168.200.15 to sip:10@192.168.200.15
    19:46:04.866 CallConf::Rejected [CM103005] Call(22) is rejected: Destination does not exist, or is not registered
    19:46:04.835 CallConf::eek:nIncoming [CM103002] Call(22): Incoming call from 105 (Ext.105) to sip:10@192.168.200.15
    19:46:03.163 MediaServerReporting::RTPSender [MS004000] Call(21) Ext.105: (ACTIVE): Can't send RTP stream to 0.0.0.0:2222 destination unreachable
    19:45:55.163 CallLegImpl::eek:nConnected [CM103001] Call(21): Created audio channel for Ext.105 (192.168.200.112:2222) with Media Server (192.168.200.15:7058)
    19:45:55.163 StratInOut::eek:nConnected [CM104005] Call(21): Setup completed for call from Ln:10000@Viatalk to Ext.105
    19:45:55.147 CallLegImpl::eek:nConnected [CM103001] Call(21): Created audio channel for Ln:10000@Viatalk (216.218.201.134:10800) with Media Server (192.168.200.15:9000)
    19:45:49.116 CallConf::eek:nProvisional [CM103003] Call(21): Ext.103 is ringing
    19:45:48.835 CallConf::eek:nIncoming [CM103002] Call(21): Incoming call from 16198088696 (Ln:10000@Viatalk) to sip:16194442200@75.22.169.234:5060

    It is still accepting 2 digits? it beeps out when i try to put in third digit.
     
  9. DaKhalli

    DaKhalli New Member

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    try to disable any local firewall running on the 3cxmachine or machine that runs any softphone.
     
  10. Denny

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    No firewall running.
     
  11. DaKhalli

    DaKhalli New Member

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    so somehow you can only type two digits, hence the 10 in the sip logs..

    Not sure why that is, maybe some1 can pick up on this one?
     
  12. Denny

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    yes. our extensions are from 100-110. I pick up call, transfer, put in any extension it beeps out right after i put in 10
     
  13. Denny

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    Found the fix.

    Fix is on the Polycom phone itself.

    Under SIP

    Local Settings

    Made sure Digitmap is cleared.

    Thanks!
     
  14. DaKhalli

    DaKhalli New Member

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    cool :D glad you solved it.
     
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