Has anyone succesfully used the Digital Receptionist

Discussion in '3CX Phone System - General' started by chrisjr, May 1, 2007.

  1. chrisjr

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    Has anyone successfully used the digital receptionist? If so could you let me know how voip provider you used, what phone, and what settings you had. I've gotten ring groups and everything else to work perfectly but whenever I add a digital receptionist it will pick my calls up and then just disconnect them. It will do this even if the calls are picked up by one of my phone. The conversation will last 10 seconds and then it will just disconnect the user.


    Chris.
     
  2. tcgmuc

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    Hi Chris,

    DR ist working in our office.

    Our Equipement for testing:

    - Windows XP Pro
    - 3CX
    - Sipgate.de as SIP-Provider
    - Linksys/CISCO Phones
    - Siemens IP-Phones

    Can you make or get any "normal" calls, or do you have the problem with the 10s every time ?

    Greethings,

    Michael.
     
  3. no1daag

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    Working fine for me

    Windows XP home :cry:
    3cx
    Wireless net
    Voip : Woize.com
    3CX Phones on laptop and three computers
    Nokia N80
     
  4. chrisjr

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    I have broadvoice, grandstream budgetone phone, and using a cingular cell phone to test the system out. I can make inbound and outbound calls just fine without it timing out after ten seconds. How do you have your digital receptionist setup?? Does it ring to a ring group?
     
  5. chrisjr

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    My server logs are below as well.... Doesn't make sense to me.

    11:25:22.343 StratLink::eek:nHangUp Call(C:6): Ln:10000@Broadovice1 hung up call; reason: BYE
    11:25:15.765 CallLegImpl::eek:nConnected Call(C:6): Created audio channel for Ext.101 (192.168.2.4:5004) with Media Server (192.168.2.6:7018)
    11:25:15.765 StratInOut::eek:nConnected Call(C:6): Setup completed for call from Ln:10000@Broadovice1 to Ext.101
    11:25:15.093 CallConf::eek:nProvisional Call(C:6): got response from 101
    11:25:09.796 MediaServerReporting::DTMFhandler from 'PhoneServer:0/MediaServer':DTMF (RTP) from 147.135.20.250:14042 arrived. in-band DTMF tone detection is turned off.
    11:25:05.687 CallLegImpl::eek:nConnected Call(C:6): Created audio channel for Ln:10000@Broadovice1 (147.135.20.250:14042) with Media Server (67.53.129.195:9000)
    11:25:05.671 CallConf::eek:nIncoming Call(C:6): Incoming call from Ln:10000@Broadovice1 to sip:2627371044@192.168.2.6
     
  6. no1daag

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    My DR have 3 chooiche
    1 to ringgroup with 2 ext
    2 to ringgroup with 3 ext
    3 to ext /voicemai

    Works fine. Use my own soundfile
    Nemas problemas

    Use only Voip provider to my 3cx
     
  7. ITWorks

    ITWorks New Member

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    Disconnect on Digital Receptionist Transfer

    I have the same problem when using the Digital Receptionist although it had been working previous to today. The line is disconnected with incoming Broadvoice IP calls seconds after transferring to the chosen extension. Everything works fine when I take the Digital Receptionist out and have calls go directly to an extension. This does not happen with PSTN calls to Grandstream Gateway.

    Windows Server 2003
    Broadvoice VSP
    Grandstream Gateway
    Cisco 7941 phones


    My logs from a failed Digital Receptionist transfer:

    14:58:12.385|.\CallConf.cpp(95)|Log2|CallControl|CallConf::eek:nIncoming:Call(C:4A): Incoming call from Ln:10001@Broadvoice to sip:26xxxx0025@10.10.30.3
    14:58:12.432|.\Call.cpp(260)|Log2|CallControl|CallLegImpl::eek:nConnected:Call(C:4A): Created audio channel for Ln:10001@Broadvoice (147.135.12.248:38692) with Media Server (10.10.30.3:9002)
    14:58:16.557|.\SLServer.cpp(303)|Log2|MediaServer|MediaServerReporting::DTMFhandler:from 'itserver1:0/MediaServer':DTMF (RTP) from 147.135.12.248:38692 arrived. in-band DTMF tone detection is turned off.
    14:58:21.369|.\CallConf.cpp(152)|Log2|CallControl|CallConf::eek:nProvisional:Call(C:4A): got response from 4414
    14:58:29.182|.\CallStrategies.cpp(721)|Log2|CallManager|StratInOut::eek:nHangUp:Call(C:4A): Call from Ln:10001@Broadvoice to 4414 has been terminated


    IVR Log:

    14:58:21.198|0c7c|c:\svn\src\ivr\vxmlivrservice\pbxmedia\IPbxMediaImpl.h(181):IvrService, IMSEndPointListenerIVR::endOfPlayedFile(), Fire '2' played
    14:58:21.198|0638|.\api\VXItelImpl.cpp(167):IvrDll, VXItelImpl::Impl::TransferBridge(), TransferBridge: 01EBBC20
    14:58:21.198|0638|.\api\VXItelImpl.cpp(120):IvrDll, VXItelImpl::Impl::IntrnTransfer(), Bridge transfer to sip:4414@127.0.0.1
    14:58:29.197|0c7c|.\PbxMedia\IPbxMediaImpl.cpp(54):IvrService, IPbxMediaImpl::stopSession(), Ivr_EndSession called
    14:58:29.197|0638|.\api\VXItelImpl.cpp(138):IvrDll, VXItelImpl::Impl::IntrnTransfer(), Error! Error of 'From_9072243326_To_1000_Menu_1000_#0' session transferring to sip:4414@127.0.0.1. Status: NEAR_END_DISCONNECT. Transfer error. Session stopped
    14:58:29.197|0638|.\api\VXItelImpl.cpp(98):IvrDll, VXItelImpl::Impl::Disconnect(), Disconnect occurred
    14:58:29.197|0c7c|.\PbxMedia\IPbxMediaImpl.cpp(35):IvrService, IPbxMediaImpl::~IPbxMediaImpl(), IPbxMediaImpl destroyed
    14:58:29.197|0c7c|.\PbxMedia\IVRServer.cpp(83):IvrService, IVRDeleteHandler(), IVRHandler:0000000A removed
     
  8. swallservices

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    From looking at your server log quoted below one thing stands out to me:

    With In-band DTMF tone detection turned off BroadVoice will be unable to pass the tones generated by key presses to your 3CX Digital Receptionist. As for why the call is terminating it's BroadVoice that's ending the call, not 3CX. There is a chance that since in-band DTMF tone detection is turned off that when BroadVoice hears a key press and knows it can't pass on the tone to 3CX it just ends the call, but that's just speculation on my part.

    Do you experience the same problems with incoming calls to a digital receptionist from a PSTN gateway or another VoIP provider other than BroadVoice? If not, you might want to contact BroadVoice to help resolve this issue.
     
  9. silentfun

    silentfun Member

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    I have set up following system

    3 extensions 100 IPPhone and 200 IPPhone and 301 (not a real phone)

    3 VoIP Providers 1001 sipgate.de and 1002 sipgate.co.uk and 1003 3.Provider

    2 DR 900 ( office open ) and 901 ( office closed ) and 903 (support request)

    if a inbound call comes from 1001 it is frowardet to 900 or 901

    both (900 and 901) DR have a greeting
    900 say we connect to the next free person and forward to 100 and/or 200
    901 say office is closed the office time is bla ...bla.. thx for calling. and forward to DR 903

    903 say If you have a emergency or a premium support contract please press "8" on your phone
    he repeat this until the caller hang up

    if the caller press "8" the DR 901 forwards the call to extension 301

    ext 301 is configured do forward the caller to a cellphone via line 1003

    there my handy is ringing
     
  10. ITWorks

    ITWorks New Member

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    disconnect after 10 seconds

    Thank you SwallServices.

    I do not believe the problem is that DTMF tones are not being detected. The digital receptionist transfers the call to the proper extension after the extension has been selected by DTMF. The call is then disconnected 10 seconds after transfer. I can see the transfer below:

    14:58:21.369|.\CallConf.cpp(152)|Log2|CallControl|CallConf::eek:nProvisional:Call(C:4A): got response from 4414
    14:58:29.182|.\CallStrategies.cpp(721)|Log2|CallManager|StratInOut::eek:nHangUp:Call(C:4A): Call from Ln:10001@Broadvoice to 4414 has been terminated

    I am sure the answer is in the IVR log below but I can not determine the cause.

    14:58:29.197|0638|.\api\VXItelImpl.cpp(138):IvrDll, VXItelImpl::Impl::IntrnTransfer(), Error! Error of 'From_5055553326_To_1000_Menu_1000_#0' session transferring to sip:4414@127.0.0.1. Status: NEAR_END_DISCONNECT. Transfer error. Session stopped
    14:58:29.197|0638|.\api\VXItelImpl.cpp(98):IvrDll, VXItelImpl::Impl::Disconnect(), Disconnect occurred


    It does not do this if the call is not transfered. It does not do this with calls coming in from the PSTN Gateway.

    The strangest thing to me is that it did work properly for a few days and just has not worked since with no conifuration changes.
     
  11. ITWorks

    ITWorks New Member

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    Digital Receptionist calls hang up after transfer

    One additional piece of information I am unable to explain.
    When calls come into the Digital Receptionist via the PSTN gateway the status shows "connected" in Line Status.
    When calls come into the Digital Receptionist via the VOIP gateway the status shows "calling" in Line Status.
     
  12. silentfun

    silentfun Member

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    i think the difference between connected & calling is if the call comes from the pstn it is connectet pysical wile a sip call is as long calling until it is established or rejectet.

    i do much testing atm and sometimes i wonder but after some thinking about i find out why most things that happens are not bugged. there is mostly a reason why.

    excuse my bad english
     
  13. ITWorks

    ITWorks New Member

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    digital receptionist transfers fail

    Chrisjr,

    Have you been able to solve this problem yet? I have spent many hours on it. It is the only thing keeping us from implementing 3CX as our school IP PBX.

    Calls coming in from Broadvoice drop when ringing an extension when transfered by digital receptionist. Never happens with calls coming in on PSTN gateway or when not using digital receptionist. Logs look the same for PSTN calls as IP calls.

    Mark
     
  14. archie

    archie Well-Known Member
    3CX Support

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    Re: Digital Receptionist calls hang up after transfer

    Yes, this explains a lot. The problem is that your VOIP gateway doesn't support re-INVITEs and/or 'Replaces:' header. Try to turn off that options in Advanced options of VOIP gateway settings - and it should help.
     
  15. ITWorks

    ITWorks New Member

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    Thanks Archie,

    This did not make a difference. The extension is ringing after DR transfer and the line status shows calling. The PSTN line shows connected when the extension is ringing after DR transfer.
    Wouldn't my 3CX machine be my VOIP gateway? I connect through a T1 and router to the Internet and BroadVoice.

    Mark
     
  16. Yuriy

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    Hi - same issue with receptionist

    I have the situation with non working receptionist extension on your Free version.
    I have downloaded a free version by the link, http://www.3cx.com/downloads/3CXPhoneSystem31.exe
    Installaed it, and did a setup:
    - configured three extensions 100,101,102.
    - two of them grouped by a RingGroup (101 and 102) with extension 800;
    - Created the receptiinist for extension 801 with the *.wav wellcome file uploaded (PCM, 8 bit, 8 KHz mono)
    > Even for pressins 1 - connect with first extension
    > Even for pressing 2 - connect with a Ring Group

    Registered all three extension, and tried to dial to Rreceptiinist

    16:39:31.890 StratInOut::eek:nCancel [CM104009] Call(9): Call from Ext.103 to 801 has been terminated
    16:39:31.875 CallConf::eek:nIncoming [CM103002] Call(9): Incoming call from 103 (Ext.103) to sip:801@x.x.x.x

    Second issue - after I've restarted 3CX* services, the call does not
    go at all:

    16:41:13.250 StratInOut::eek:nHangUp [CM104007] Call(10): Call from Ext.103 to 101 has been terminated by Ext.103; cause: CANCEL; from IP:x.x.x.x
    16:41:08.515 gt;>:Illegal message rejected: ParseBuffer.cxx:948, Parse failed unexpected eof in context: From
    16:41:08.390 CallConf::eek:nIncoming [CM103002] Call(10): Incoming call from 103 (Ext.103) to sip:101@x.x.x.x
     
  17. Ralph

    Ralph Member

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    Hi

    We are using the following:

    3CX Server:
    OS: Windows Server 2003 SP2
    RAM: 1 Gig
    CPU: P4 3 Ghz

    Phones:
    Polycom Soundpoint IP 501s
    Grandestream GXP 2000
    Grandstream Handytone 486

    All phones are on the same subnet except the Grandstream Handytone which is connected via a VPN from a satalite office.

    Digital receptionist (we have a total of 7 set up) all seem to work fine.

    Take care
     
  18. 5qg4

    5qg4 Active Member

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    Hi Yuriy,

    Please enable the log level to "Verbose (only used for debugging purposes)". Restart all 3cx services or restart 3cx box. Then post the log here again.
     
  19. Ralph

    Ralph Member

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    Hi

    Hi Yuriy,

    You mentioned that you set up extensions and a ring group but not that you set up a digital receptioist.

    Did you set up a digital receptionist and did you configure your incomming lines to be answered by the digital receptionist?

    Take care,
     

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