Transfers not happening on Polycom 501

Discussion in '3CX Phone System - General' started by porscheman7, Oct 5, 2007.

  1. porscheman7

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    I am using the latest build of 3CX and have 10 Polycom 501 phones. All calls ring to extention 110. If no answer there, ring to Digital Receptionist.
    when calls answered at 110, or any other phone, are transferred, the following log entries show up.

    11:47:57.703 StratInOut::eek:nHangUp [CM104007] Call(793): Call from Ext.110 to 113 has been terminated by Ext.110; cause: CANCEL; from IP:192.168.0.8
    11:47:57.663 StratLink::eek:nHangUp [CM104001] Call(792): Ext.110 hung up call; cause: BYE; from IP:192.168.0.8
    11:47:57.513 CallStrategy::Transfer [CM104000] Call(792): transfer failed
    11:47:57.513 StratTransfer::initialize [CM008004] Call(792): Cannot build transfer target by DBCfg
    11:47:57.513 StratLink::Transfer [CM104002] Call(792): transfers Ln:10004@Binary Telecom Main from Ext.110 to sip:113@192.168.0.20
    11:47:56.401 CallConf::eek:nProvisional [CM103003] Call(793): Ext.113 is ringing
    11:47:56.010 CallConf::eek:nIncoming [CM103002] Call(793): Incoming call from 110 (Ext.110) to sip:113@192.168.0.20
    11:47:53.196 MediaServerReporting::RTPSender [MS004000] Call(792) Ext.110: (ACTIVE): Can't send RTP stream to 0.0.0.0:2266 destination unreachable
    11:47:39.467 CallLegImpl::eek:nConnected [CM103001] Call(792): Created audio channel for Ext.110 (192.168.0.8:2266) with Media Server (192.168.0.20:7172)
    11:47:39.447 StratInOut::eek:nConnected [CM104005] Call(792): Setup completed for call from Ln:10004@Binary Telecom Main to Ext.110
    11:47:39.447 CallLegImpl::eek:nConnected [CM103001] Call(792): Created audio channel for Ln:10004@Binary Telecom Main :)58030) with Media Server (206.207.108.227:9266)
    11:47:36.372 CallConf::eek:nProvisional [CM103003] Call(792): Ext.110 is ringing
    11:47:35.992 CallConf::eek:nIncoming [CM103002] Call(792): Incoming call from 2088510438 (Ln:10004@Binary Telecom Main) to sip:12084170153@69.30.45.229

    All Lines from VOIP Provider have PBX delivers audio checked, and all extentions have Bind to Media server checked.

    Phones have latest version of SIP files installed (2.1.2)

    Any Ideas?
    Thanks
     
  2. porscheman7

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    Fixed it

    Changed SIP ID in Other options in extension management to extension number, Uncheck support Re-invite and support 'replace'header boxes.

    blind and attended transfers now work.

    YEA!!!!!
     

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