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McFone

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Posts posted by McFone

  1. Park Orbit

    in General Setup

    Posted

    Hi,

     

    how can we activate this blind transfer without connect to the second caller just by pressing a button.

    The secretary does not want to interrupt the existing call.

     

    Maybe for the next version a good feature "forward when busy to a park orbit" with DND it is working.

     

    thanks

  3. Redirect

    in General Setup

    Posted

    Hi,

     

    we have a problem with the redirect button

     

    Redirect: This mode allows you to redirect calls to a predefined extension or number. To use this mode, enter the redirection target into the parameter field. Pressing the button activates redirection and lights the LED. Pressing the button deactivates redirection. This mode is useful when an executive assistant should take over calls.

     

    Pressing the button activate or deactivate the redirect and the display shows the redirection.

    Your problem is the button have not lights only in the display.

    We use the PNP function and the button are custom not a templat.

     

    We use a snom 720 and 5.0.10a snomONE

     

    thanks

     

  4. Park Orbit

    in General Setup

    Posted

    Hi,

    we try to forward calls when busy to a Park Orbit.
    The idea is the secretary is on the phone and have a second call and through the busy flag forward this call to a park orbit and a other employee can pickup this call. We try this and the call is not forwarding to a park orbit account.

    Any idea what is wrong or we can forward a call to a park orbit ?

    thanks

     

     

  5. User Portal

    in Web Interface

    Posted

    Hi,

     

    we use 5.0.10a and the account is green in the user portal.

    For testing we have a snom 720 and one snom 760 registred to one account and try the click2dial with the second registred phone snom 760.

    Maybe this is the reason two registration with different snom phone types on one account.

     

    I will check click2dial on a free account with only one phone is this working

     

    thanks

     

    P.S.: Any idea to hide the extensions etc in the Web GUI

  6. User Portal

    in Web Interface

    Posted

    Hi,

     

    the user portal is very nice ;-)

    Our issues with this webinterface is

     

    1.) Refresh

    Calls not updatet in the interface until we refresh the browser manually the extension is direct updatet (RED).

    After a manually refresh the call is showing and for a test we canncel this call all is direct updatet extension (green) and the call is gone.

    Why we need to display a call a manually refresh

    We know we need websocket browser IE10, Chrome etc. we use chrome

     

    2.) Click2Dial

    We push the button you can hear a short signal there is no number shown in the display after 2-3 sec later the call is signaling to the dialing fone. The problem is we can take on this call from the dialing fone and we are connected on the extension we can not hear anything.

     

    Call this fone direct no problems (2 ways) the PBX is install on the server in the wan.

     

    3.) Hide Extension

    How we can hide extensions in the interface

    We try under extension monitoring Permissions to monitor this account no change in the webinterface

     

    thanks

  7. second call transfer

    in Call Treatment

    Posted

    Hi,

     

    we find not the right way to handle this call flow

    The secretary is talking and get a second call this call she want to forwarded directly to a extension without the first call to put on hold.

     

    Any idea how we can handle this maybe via the webinterface ?

     

    thanks

     

     

     

  8. Request time out

    in General Setup

    Posted

    Sounds like a problem with the routing of the ACK request. What service provider are you using? Does this service provider has a "session border controller"? If not, then you must operate the PBX on a public IP address. If you like, filter in the LOG for the IP address of the service provider and attach the SIP packets here; then we can take a look.

     

    send a wireshark trace via PM

     

    thanks

  9. Voicemail Announcement only

    in Mailbox

    Posted

    I still don't 100 % understand the use case. So every user wants to have the ability to redirect a call into a IVR announcement? So lets say for example when you have some explanations that you want to play back to explain something you would blind transfer the call into one of the pre-recorded announcements?

     

    Normaly we have a set of IVR's for different cases Meeting, Out of Office etc and all the employee link to this IVR with a button on the phone

    The director from the company want 3 independent announcement for every employee this is the requirement

     

    customer is king

  10. DND deactivation status update

    in General Setup

    Posted

    [/size]

     

     

    Oh that! Yea there is a problem with the phones that dial the number at teh same time when using the bespoken feature synchronization. Newer firmware versions have that problem fixed, however are not ready for release yet. The calls eventually time out, and then it could be that the status is cahnged some time later.

     

    all the phones have 8.7.3.19 !

    the only thing we can do is to wait for the new firmware is this right :-(

     

    thanks

  11. Transfer on Display

    in General Setup

    Posted

    Hi,

     

    the display on the snom 720 show TRANSFER for incoming calls how work this ?

    Press the button Transfer and dial a extension or press the BLF button is not working.

     

    This function is supported by the SnomOne ?

     

    thanks

  12. Info Display

    in General Setup

    Posted

    Hi,

     

    the snom 720 have a INFO button on the display for Information around the phone for call forward time etc. we want to control this INFO field maybe disable or delete the messaging etc.

    The reason is the update from 5.08 to 5.09 gives a error the message in the INFO was prov server failed etc.

    Next day we have a lot of calls why the phone have a error bla bla.

     

    snomONE can control this behavior

     

    The bug with the redirction hope is fix in the next version.

     

    thanks

  13. DND deactivation status update

    in General Setup

    Posted

    Hi,

     

    we test the DND function with a snom 870 the same result the snom 870 have also a PNP config.

    Activate is direct and deactivate takes a long time to refresh, is this a snomone problem or a snom firmware and the must open direct by snom a ticket.

     

     

    The PBX send this information via email

     

    The call between sip:*79@pbx.awo.de;user=phone and sip:41@pbx.awo.de has been disconnected because media session was not established (source=xxxxxxxxxx:3326)

    The call between sip:*78@pbx.awo.de;user=phone and sip:41@pbx.awo.de has been disconnected because media session was not established (source=xxxxxxxxxx:3326)

     

    2013/5/9 20:58:27 Tx: tls:xxxxxxxxxxx:3326 (307 bytes)

    SIP/2.0 100 Trying

    Via: SIP/2.0/TLS 192.168.178.24:3326;branch=z9hG4bK-x32h21wzvwx4;rport=3326;received=xxxxxxxxxx

    From: "AWO Konferenzraum" <sip:41@pbx.awo.de>;tag=2ax4aofm16

    To: <sip:*78@pbx.awo.de;user=phone>;tag=40e1a2c850

    Call-ID: d3f18b51114b-huu01m607kxr

    CSeq: 1 INVITE

    Content-Length: 0

     

    2013/5/9 20:58:27 Tx: tls:xxxxxxxxxxx:3326 (894 bytes)

    SIP/2.0 200 Ok

    Via: SIP/2.0/TLS 192.168.178.24:3326;branch=z9hG4bK-x32h21wzvwx4;rport=3326;received=xxxxxxxxxx

    From: "AWO Konferenzraum" <sip:41@pbx.awo.de>;tag=2ax4aofm16

    To: <sip:*78@pbx.awo.de;user=phone>;tag=40e1a2c850

    Call-ID: d3f18b51114b-huu01m607kxr

    CSeq: 1 INVITE

    Contact: <sip:41@xxxxxxxxx:5061;transport=tls>

    Supported: 100rel, replaces, norefersub

    Allow-Events: refer

    Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

    Accept: application/sdp

    User-Agent: snomONE/5.0.8

    Content-Type: application/sdp

    Content-Length: 333

     

    v=0

    o=- 1975364849 1975364849 IN IP4 xxxxxxxx

    s=-

    c=IN IP4 xxxxxxxxxx

    t=0 0

    m=audio 57078 RTP/AVP 8 101

    a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:zo1r7tsVBKAONiXFzvWwm2S7T31WvD2IC1f8YTq4

    a=rtpmap:8 PCMA/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-16

    a=ptime:20

    a=rtcp-xr:rcvr-rtt=all voip-metrics

    a=recvonly

    2013/5/9 20:58:27 Rx: tls:xxxxxxxxxx:3326 (433 bytes)

    ACK sip:41@xxxxxxxxxxxx:5061;transport=tls SIP/2.0

    Via: SIP/2.0/TLS 192.168.178.24:3326;branch=z9hG4bK-fd9n70gy2vqv;rport

    From: "AWO Konferenzraum" <sip:41@pbx.awo.de>;tag=2ax4aofm16

    To: <sip:*78@pbx.awo.de;user=phone>;tag=40e1a2c850

    Call-ID: d3f18b51114b-huu01m607kxr

    CSeq: 1 ACK

    Max-Forwards: 70

    Contact: <sip:41@192.168.178.24:3326;transport=tls;line=tjdf0bu5>;reg-id=1

    Proxy-Require: buttons-snom870

    Content-Length: 0

     

     

    2013/5/9 20:58:12 Tx: tls:xxxxxxxxxxxxx:3326 (307 bytes)

    SIP/2.0 100 Trying

    Via: SIP/2.0/TLS 192.168.178.24:3326;branch=z9hG4bK-l06jawifprit;rport=3326;received=xxxxxxxxxxxxx

    From: "AWO Konferenzraum" <sip:41@pbx.awo.de>;tag=yaxlta276t

    To: <sip:*79@pbx.awo.de;user=phone>;tag=fa7272e543

    Call-ID: c4f18b5197a4-xqivq2gebv5x

    CSeq: 1 INVITE

    Content-Length: 0

     

    2013/5/9 20:58:12 Tx: tls:xxxxxxxxxxxxx:3326 (894 bytes)

    SIP/2.0 200 Ok

    Via: SIP/2.0/TLS 192.168.178.24:3326;branch=z9hG4bK-l06jawifprit;rport=3326;received=xxxxxxxxxxxxx

    From: "AWO Konferenzraum" <sip:41@pbx.awo.de>;tag=yaxlta276t

    To: <sip:*79@pbx.awo.de;user=phone>;tag=fa7272e543

    Call-ID: c4f18b5197a4-xqivq2gebv5x

    CSeq: 1 INVITE

    Contact: <sip:41@xxxxxxxxxxxxx:5061;transport=tls>

    Supported: 100rel, replaces, norefersub

    Allow-Events: refer

    Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

    Accept: application/sdp

    User-Agent: snomONE/5.0.8

    Content-Type: application/sdp

    Content-Length: 333

     

    v=0

    o=- 1284771858 1284771858 IN IP4 xxxxxxxxxxxxx

    s=-

    c=IN IP4 xxxxxxxxxxxxx

    t=0 0

    m=audio 63032 RTP/AVP 8 101

    a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:vya3Cx1y/JkuKznovqkIfCmaSTQo31ZQir61D23B

    a=rtpmap:8 PCMA/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-16

    a=ptime:20

    a=rtcp-xr:rcvr-rtt=all voip-metrics

    a=recvonly

    2013/5/9 20:58:13 Rx: tls:xxxxxxxxxxxxx:3326 (433 bytes)

    ACK sip:41@xxxxxxxxxxxxx:5061;transport=tls SIP/2.0

    Via: SIP/2.0/TLS 192.168.178.24:3326;branch=z9hG4bK-0n1zfqsdgi5y;rport

    From: "AWO Konferenzraum" <sip:41@pbx.awo.de>;tag=yaxlta276t

    To: <sip:*79@pbx.awo.de;user=phone>;tag=fa7272e543

    Call-ID: c4f18b5197a4-xqivq2gebv5x

    CSeq: 1 ACK

    Max-Forwards: 70

    Contact: <sip:41@192.168.178.24:3326;transport=tls;line=tjdf0bu5>;reg-id=1

    Proxy-Require: buttons-snom870

    Content-Length: 0

  14. Voicemail Announcement only

    in Mailbox

    Posted

    I would maybe try recording these greetings on the extension level shown here and if the user is off to lunch they would have to call into the users VM and choose that specific greeting or they can choose it from the User portal as well.

     

    thanks for answer

    the company want only one voicebox for the director and the setting in mailbox to max message to "0" for the other extensions brings the message voicebox is full. Maybe you know how we can set for each extension the record time for a voicemail?

    In the domain settings i can to this only for all extensions.

     

    As a workaround we can set the time of record for a voicebox of 1 min or less.

     

    thanks

  15. Voicemail Announcement only

    in Mailbox

    Posted

    The requirements for a pbx of the new company is very low.

    This company need only very basic features without voicemail or mobile integration etc.

    One thing that is important is a AA and personal announcement.

     

    The company have 10 employees and for every one the company need 3 personal announcements and 2 for the AA.

    For 10 accounts we have only 10 IVR's in the package so the company must pay 499$ + 2200$ (22*99$ IVR) = 2699$ for a very small PBX !!

    I can not sold this expensive solution to this company.

     

    for me looks like the price for ivr should be redesigned in many cases you use the IVR announcement before notification etc.

     

    maybe i have understand the ivr or the license not right.

  18. Send to extension

    in IVR Node Setup

    Posted

    Hi,

     

    for a support hotline we have a premium service.

    In a webinterface your customer can create a string for this service to input this combination to the IVR.

    This string we write back to snomone this solution works perfect.

     

    What we need is to know the expression how we can tranfer a caller to a second IVR or extension when this input of the DTMF is not matching to announcement a message etc.

     

    thanks

    post-18921-0-11803800-1368102270_thumb.jpg

  19. Voicemail Announcement only

    in Mailbox

    Posted

    Hmm. You could limit the number of messages in the voicemail to zero. That should have the same effect like a full mailbox.

     

    Hi,

     

    we try to set the voicemail to "0" and the system gives back a announcement "mailbox are full" and play not our audio files for the greetings.

    Maybe we need to change more parameters or we can change the xml file of snomone to have a workaround for this.

    We need this function urgently !

     

    thanks

  21. DND deactivation status update

    in General Setup

    Posted

    Hi,

     

    if we set DND on the phone this extension is immediately marked as busy.

    The DND deactivation status on the phone takes about 120 sec before this update is marked on the phone and also on the other extensions.

     

    Why the deactivation status is not immediately on the phones we use snom 720 ?

     

    thanks

  22. Request time out

    in General Setup

    Posted

    Hi,

     

    since we have updated to snomone v5 we have a request time out after 30-32 sec for outbound calls

    The call is normal ringing and after this time we have a message request time out.

     

    How we can change this?

     

     

    thanks

  25. ACD cell phone integration

    in Using Cell Phones

    Posted

    Hi,

     

    the cell phone can truly act as an extension of the PBX with new menu options, ability to transfer the call on the cell phone to another extension (with a ## option), bridge/conference calls, dial another number, stay connected to PBX to initiate new call even after the remote party of the previous call hangs up etc.

     

    Works fine with extensions and Huntgroups but not for agents (ACD) with a redirect to a mobile phone.

    ## option is not available

     

    thanks

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