Jump to content

Hebel IT

Members
  • Posts

    6
  • Joined

  • Last visited

Posts posted by Hebel IT

  1. Ok, it seems I'm still making a fundamental mistake:

     

    I'm interested only in modifying the FROM address in the phone display (not at all for routing purposes), so I'm using the

    expression "!([0-9]*)!+\1!f!50" in the field "Send call to extension" in the configuration of the trunk.

     

    [8] 2012/06/15 15:28:38:	Call from a trunk 4
    [8] 2012/06/15 15:28:38:	Trunk alonia.ro@myPbxDomain has country code not set, area code not set
    [8] 2012/06/15 15:28:38:	To is <sip:40373711111@85.204.232.10;user=phone>, user 0, domain 1
    [8] 2012/06/15 15:28:38:	Trying to match number 40373722222 with ERE ([0-9]*)
    [8] 2012/06/15 15:28:38:	Send call to extension ERE returned +40373722222
    [8] 2012/06/15 15:28:38:	User not found for account +40373722222 in domain myPbxDomain, trying to send the call to default account 50
    [5] 2012/06/15 15:28:38:	Domain trunk alonia.ro@myPbxDomain sends call to 50 in domain myPbxDomain

     

    As I can see from the log the regular expression prepended the plus-sign correctly but the FROM number displayed on my phone continues to be "40373722222".

     

    Is the field "Send call to extension" really the right thing to achieve what I want?

     

    Best regards

     

     

    Florian

  2. Sorry to bother you again, but unfortunately I didn't succeed to get it working.

     

    Here is my requirement more in detail:

     

    I have the following incoming call, which I want to send always to extension 50 (a hunt group).

    In addition to that I want to see the caller's number on the phone as +40373722222 instead of 40373722222 in order to be able to use the entry from the callers' list to call back.

     

    INVITE sip:40373711111@192.168.100.5:5060;transport=udp;line=a87ff679 SIP/2.0
    Record-Route: <sip:85.204.232.10;ftag=c82c163e0942f84b89be068a1b2fa668;lr>
    Via: SIP/2.0/UDP 85.204.232.10;branch=z9hG4bK6202.acb0fdb8515ae4f906c5fc607f3bb810.0
    Via: SIP/2.0/UDP 85.204.232.10:5061;branch=z9hG4bKb971741b4a3b11e74efee46b38ad8806;rport=5061
    Max-Forwards: 16
    From: <sip:40373722222@85.204.232.10>;tag=c82c163e0942f84b89be068a1b2fa668
    To: <sip:40373711111@85.204.232.10>
    Call-ID: 49795A3309CECECA10237350FE1E11FC4564B5DE
    CSeq: 200 INVITE
    Contact: Anonymous <sip:85.204.232.10:5061>
    Expires: 300
    User-Agent: Sippy
    cisco-GUID: 554731087-3059945953-3006070804-571693236
    h323-conf-id: 554731087-3059945953-3006070804-571693236
    Content-disposition: session
    Content-Type: application/sdp
    Content-Length: 266

     

    Thanks in advance for your help.

     

    Best regards

     

    Florian

  3. Ciao ragazzi,

     

    Something for the next upcoming versions is probably a dial plan for inbound purposes.

     

    I would urgently need this "inbound dial plan" because all my customers have trunks from at least two countries and I have to provide the callers' numbers in the international format.

     

    What's the planned schedule on this topic?

     

    Thanks in advance.

     

    Tanti saluti

     

    Florian

  4. Hi all,

     

    yesterday I got my brand-new "snom ONE SOHO". Given that I need to support several

    different languages I mounted an ext2 filesystem on an external usb-stick (/etc/fstab),

    moved the audio_de, audio_it, audio_sp & audio_uk to that fs and created symlinks

    in /usr/local/snomONE, which point to /mnt/usb/audio_de, etc. ...

     

    Unfortunately when I restart the service the "external" languages are not found any

    more.

     

    What am I doing wrong?

     

    Thanks in advance for your help.

     

    Best regards

     

    Florian

×
×
  • Create New...