Trasfer calls from extension to another extension

Discussion in '3CX Phone System - General' started by costellazione, Apr 7, 2007.

  1. costellazione

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    Hello,
    could somebody explain us how we can trasfer calls from extension to another extension using regular telephony keys?

    There is a table of fast digit to do the most useful function by telephony keys?

    Thank you in advance if you would help me and thousand thanks to Nick Galea for his performances.

    ciao
    Costellazione
     
  2. Nick Galea

    Nick Galea Site Admin

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    Hi Costellazione,

    Actually each phone should have a transfer button/function. For example the Grandstream GXP 2000 has a transfer button to do a blind transfer. With attended call transfer you dial the other extension first on the second line of the phone and then connect them.

    With SJphone (software SIP Phone) there should be a button to transfer calls. Note that 3Cx Phone does not have a transfer function yet (we are working on it, focusing on phone system first) and X-lite requires a paid upgrade to do a transfer.

    Hope this helps. And thank you for the heads up! :) Actually its the development and testing team who should get the credit :)
     
  3. costellazione

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    Thank you, Nick.

    My question was made just to understand if your 3cx pbx server accept some type of special input as other pbx do.
    In example, Asterisk accept special digits (##) to transfer calls from any kind of telephon to other extension.

    This simple (I know this is not simple programming), if not already exist, would be implemented because very useful.
    This is my thinking. :)

    ciao
    Costellazione
     
  4. Nick Galea

    Nick Galea Site Admin

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    Hi Costellazione,

    Makes sense..... At this point we do not have such a function but it makes sense and we will definitely consider it for future builds....!

    Thanks for the suggestion!
     
  5. pkothe

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    Help with Transfer

    I am able to do an attended transfer but have not figured out how to do a blind transfer with a Linksys 942 or 962. Any suggestions?
     
  6. pirkaiia

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    I have a Linksys SPA962 and a Cisco 7912G and when you want to transfer a call on the SPA962 you have in the lower right corner a softkey with the tekt transfer. When you press transfer, you get a dialtone, dial an other extension nr, wait for connection, en press transfer again to compleet the transfer, both partys have now a connection and your out of it.

    see log:
    23:57:44.453 StratLink::eek:nHangUp [CM104001] Call(30): Ext.114 hung up call; cause: BYE; from IP:172.16.0.254
    23:57:40.015 StratLink::eek:nHangUp [CM004001] Call(30): HangUp from wrong leg has arrived. Leg will be removed from call
    23:57:36.015 StratTransfer::finishTransfer [CM108005] Call(30): Transfer finished, Ext.113 is connected to Ext.114
    23:57:35.796 StratLink::Transfer [CM104002] Call(30): transfers Ext.113 from Ext.200 to sip:114@172.16.0.200
    23:57:31.437 CallLegImpl::eek:nConnected [CM103001] Call(31): Created audio channel for Ext.114 (172.16.0.254:16446) with Media Server (172.16.0.200:7116)
    23:57:31.437 StratInOut::eek:nConnected [CM104005] Call(31): Setup completed for call from Ext.200 to Ext.114
    23:57:31.437 CallLegImpl::eek:nConnected [CM103001] Call(31): Created audio channel for Ext.200 (172.16.0.199:16470) with Media Server (172.16.0.200:7114)
    23:57:25.765 CallConf::eek:nProvisional [CM103003] Call(31): Ext.114 is ringing
    23:57:25.671 CallConf::eek:nIncoming [CM103002] Call(31): Incoming call from 200 (Ext.200) to sip:114@172.16.0.200
    23:57:20.562 MediaServerReporting::RTPSender [MS004000] Call(30) Ext.200: (ACTIVE): Can't send RTP stream to 0.0.0.0:16468 destination unreachable
    23:57:16.671 CallLegImpl::eek:nConnected [CM103001] Call(30): Created audio channel for Ext.200 (172.16.0.199:16468) with Media Server (172.16.0.200:7112)
    23:57:16.671 StratInOut::eek:nConnected [CM104005] Call(30): Setup completed for call from Ext.113 to Ext.200
    23:57:16.671 CallLegImpl::eek:nConnected [CM103001] Call(30): Created audio channel for Ext.113 (172.16.0.254:16444) with Media Server (172.16.0.200:7110)
    23:57:13.890 CallConf::eek:nProvisional [CM103003] Call(30): Ext.200 is ringing
    23:57:13.781 CallConf::eek:nIncoming [CM103002] Call(30): Incoming call from 113 (Ext.113) to sip:200@172.16.0.200


    when I look on the web page op the phone, tab region, under "Vertical Service Activation Codes", you see the "Blind Transfer Code: = *98"parameter. Maybee it wil work alsoo if you dial *98 to transfer the call, I haven't tried yet.

    For the Cisco 7912G you have to press in the lower right corner More to come to the softkey Transfer.

    I hope it wil work for you.

    Not bad he for a newbee.

    Pirkaiia
     
  7. pirkaiia

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    I just tried to tranfer a call with *98, <dialtone>, dial extension nr., and it works as well.

    Pirkaiia
     
  8. pkothe

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    I found another way to do it also

    On the Linksys phone there is a Transfer Line soft key that comes up when you do this.

    Answer call and put it on hold
    Pick up spare extension and dial other extension
    While it is ringing there is a Xfer Line button and you hit that and away it goes.

    Thanks,

    Paul
     
  9. Urbok

    Urbok New Member

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    well but this ## function well be posted on future beta releases?
    or it's already programmed in the actual beta version?

    let me know because with voice gateways as Grandstream HT386 I'm not able to complete transfers between extensions.
     
  10. Urbok

    Urbok New Member

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    UP!

    Please dont force me to installa asterisk instead of this super PBX.
    noway to transfer calls from voice gateways as grandstream HT386??

    sigh
     
  11. pkothe

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    Grandstream Gateway Transfer

    Can you detail the procedure and what type of phone you are using? I am trying to think back and I seem to remember having to change a parameter in the Linksys phone to allow Blind Transfers. Also look at your settings on your PBX under the gateway config - other options make sure you have the "PBX delivers Audio" checked and the "Gateway is External" unchecked. I am assumoing your gateway and your PBX machine are on the same subnet. If you make sure your RTP stream is flowing thru the PBX then it will have control of the call so when you transfer to another extension it should stay the hub of the call.

    Hope this helps,

    Paul
     
  12. Urbok

    Urbok New Member

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    I use an handytone 386 with siemens cordless analog phones.
    during conversation when I push the FLASH key I dont receive a special tone... simply nothing happens.

    i've tried to change on cfg of handytone the way used to send DTMF to the server bue nothing no way to transfer a call.
    I've tried to set the flash key as a DTMF but nothing.

    In asterisk the PBX intercept the ## characters and wait to the number of the extension to complete the call transfer. ( a standard way to transfer from extension to extension regardless of hardware.)

    this is a terrible limitation.... I've asked also to grandstream if there is something that I can try on the handytone configuration but noone replied me.

    SIGH :-((


    Here is the config of FXS port 1: (as the port 2 naturally)



    Use DNS SRV: Yes
    User ID is phone number: Yes
    SIP Registration: Yes
    Unregister On Reboot: Yes
    Register Expiration: (in minutes. default 1 hour, max 45 days)
    local SIP port: (default 5060)
    local RTP port: (1024-65535, default 5004)
    Use random port: No
    SIP Registration Failure Retry Wait Time: (in seconds. Between 1-3600, default is 20)
    DTMF Payload Type:
    Send DTMF: in-audio [X] via RTP (RFC2833) via SIP INFO
    Send Flash Event: [X] No Yes (Flash will be sent as a DTMF event if set to Yes)
    Enable Call Features: No [X] Yes
    (if yes, call features using star codes will be supported locally)
    Use Bell-style
    3-way Conference: [X] No Yes (if Yes, *23 will be disabled)
    Offhook Auto-Dial: (User ID/extension to dial automatically when offhook)
    Proxy-Require:
    Disable Call-Waiting: [X] No Yes
    Disable Call Waiting Caller ID (CWCID): [X] No Yes (If set Yes, users cannot see Call-Waiting Caller-ID)
    NAT Traversal (STUN): [X] No Yes
    No Key Entry Timeout: (in seconds, default is 4 seconds)

    Preferred Vocoder:
    (in listed order) choice 1: current setting is " PCMU" G.723.1 G.729A/B PCMU PCMA iLBC G.726-32
    choice 2: current setting is " PCMA" G.723.1 G.729A/B PCMU PCMA iLBC G.726-32
    choice 3: current setting is " G729" G.723.1 G.729A/B PCMU PCMA iLBC G.726-32
    choice 4: current setting is " G723" G.723.1 G.729A/B PCMU PCMA iLBC G.726-32
    choice 5: current setting is " G726-32" G.723.1 G.729A/B PCMU PCMA iLBC G.726-32
    choice 6: current setting is " iLBC" G.723.1 G.729A/B PCMU PCMA iLBC G.726-32
    Voice Frames per TX: 2 (up to 10/20/32/64 for G711/G726/G723/other codecs respectively)
    G723 rate: [X] 6.3kbps encoding rate 5.3kbps encoding rate
    iLBC frame size: [X] 20ms 30ms
    iLBC payload type: 97 (between 96 and 127, default is 97)
    Silence Suppression: [X] No Yes
    Fax Mode: T.38 (Auto Detect) Pass-Through
    Early Dial: No Yes (use "Yes" only if proxy supports 484 response)
    Allow incoming SIP messages
    from SIP proxy only: [X] No Yes
    Allow outgoing call without Registration: [X] No Yes
    Dial Plan Prefix: (this prefix string is added to each dialed number)
    Use # as Dial Key: No [X] Yes (if set to Yes, "#" will function as the Dial key)
    SUBSCRIBE for MWI: [X] No, do not send SUBSCRIBE for Message Waiting Indication
    Send Anonymous: [X] No Yes (caller ID will be blocked if set to Yes)
    Lock keypad update: [X] No Yes (configuration update via keypad is disabled if set to Yes)
    Refer-To Uses Target Contact [X] No Yes
    Special Feature: [X] Standard CBCOM RNK
     
  13. nickybrg

    nickybrg Well-Known Member
    3CX Staff

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    Every Phone and manufacturer have different codes to affect blind and attended transfers.

    We have a list to affect these transfers but the list relates to the major phones that are supported by the 3CX Phone System. Any queries on how to affect these transfers please ask.
    --
    Example the cisco IP Phone 7940 series uses the following procedures to effect blind and attended transfers :

    Blind:
    Current conversation
    press more button (current conversation goes on hold)
    BXFER (for blind trans)
    the extension number and finally
    Dail button
    --(the first call is blindly transfered to the second extension dialled without conversation in between)

    Attended:
    Current conversation
    Press "More" - Current conversation goes on hold.
    Transfer button (a new line is created)
    dial the extension number you would like to transfer too
    Talk with the second user
    Decide to effect the transfer and press the TRANS button.
    --The original call and the last call will connect together.

    Grandstream GXP 2000

    Blind transfer:
    --Trans
    --enter Extension Number you want call to be transfered to
    --SEND button

    Attended transfer:

    Current incoming call
    --Hold button
    --press L2 (line 2)
    --enter extension number
    --Talk to user of this extension number - decide to affect transfer - ok
    --Press on the following buttons in order - TRANS and L1
    L1 will automatically connect to L2 and the users will talk to each other.
     

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