2 problems with the server - 3cx

Discussion in '3CX Phone System - General' started by hagesol, Mar 30, 2007.

Thread Status:
Not open for further replies.
  1. hagesol

    Joined:
    Mar 26, 2007
    Messages:
    22
    Likes Received:
    0
    hi

    I have 2 problems...

    1)

    I can call from 100 to 101 and from 101 to 100 (Extensions) but not out with VOIP Provider (sip). (I use the X-lite softwarephone for calling, I have tried other softwarephones but then I can not get any connections and sound.) I got this Message in the logg:

    11:03:09.546 CallConf::Rejected Call (C:1C) is rejected: Calling is failed (unspecified)

    11:03:09.546 StratInOut::eek:nCancel Call from Ext.100 to 0477258xxxx has been terminated; reason: Calling is failed (unspecified)

    11:03:00.671 CallConf::eek:nIncoming Incoming call from Ext.100 to sip:00477258xxxx@192.168.0.174

    I have tried and tried to understand, but I have not more ideas.

    In the beginning I had som stunt server problems, but that is ok now. I got not more stunt server problems.

    When I call a inter-call I have sound and the other person can hear me. (100-101 and 101-100)

    I have also set up a Digital Receptionist and I hear the voice from the reseptionist, but I can not use the button (1 for 100 and 0 for 101).

    Sorry for my english...
    I hope you understand my problem?
    I hope someone out there can help me...

    lars
    (hagesol)
     
  2. Costas3CX

    Costas3CX New Member

    Joined:
    Jan 23, 2007
    Messages:
    217
    Likes Received:
    0
    What is your Voip Provider?
    What version of PBX are you running?
    what version of xlite are you running?
     
  3. Anonymous

    Anonymous Guest

    Ok for your two problems I will try to fix one :)
    It looks like a DTMF issue, (you can do a search on the forum) you will find that 3cxsupport mentioned that codecs G729 will not work with DTMF, hence you have to set this to G711 (u/a).

    If you have that than you can do ONE of the following options.

    1. (with no firewall)
    For each extension 100 and 101 configure in the advanced options Device is External. I say for no firewall, because it will use the ports 9000-9003 when you select this and that will trigger the use of an external IP type scenario.

    2. (with a firewall)
    For each extension 100 and 101 configure in the advance options Bind to Media Server. This will use the internal ports 7000 - 7500 and internal ip address type scenario.

    Only do one or the other, not both.

    I will try to have a go at your other problem, but it is a bit confusing the way you lay it out.
    Although you say you do not use a VOIP, the log shows otherwise (well what I understand from it. It looks like the number is routed to an external number.

    Anyway have a look, another tip, becareful in posting IP addesses and phone numbers on forums. They are not to bad here, but google brings up some posts of this forum when searching. Just a heads up for ya.

    Henk.
     
  4. hagesol

    Joined:
    Mar 26, 2007
    Messages:
    22
    Likes Received:
    0
     
  5. hagesol

    Joined:
    Mar 26, 2007
    Messages:
    22
    Likes Received:
    0
    !!!!

    Thanks ItFarmer,,, now is the Digital Receptionist ok,,,

    but to the 3CX TEAM - do you have a solusion to the other question?

    lars
     
  6. Anonymous

    Anonymous Guest

    What is your configuration on 101.

    Base upon what I see 101 is forwarded to an external number, is that the case?

    Are you trying to dial from internal to external? Or only internal to internal?
     
  7. hagesol

    Joined:
    Mar 26, 2007
    Messages:
    22
    Likes Received:
    0
    hi again

    I trying to dial from internal to external, internal to internal (intranet) is ok. Just to be sure that is not a firewall problem in the router, I have made a user (103) outside the intranet. We can call eachother.

    The configuration for all users is the same (100,101 and the new 103):

    Device is External - no
    Bind to Media Server - yes
    Supports Re-Invite - yes
    Supports 'Replaces' header - yes

    I have set up the server like this: when all users call eachother they come to the Digital Receptionist - this is function now after I get some info from you.

    I have also made a Outbound Rules that do this: place 0047 before the number the users dial, and what provider (sip) I will use.

    When I call outside (external I get the message in the logg that I have posted earlyer. When I call from skype to my server (sip - phonenumber from http://www.phonzo.com) I get a busy tone.

    This is my problem....

    Lars
     
  8. Anonymous

    Anonymous Guest

    Can you check if you actually register with your VSP?

    Looks like all the things are in place for 3cx but the moment it leaves your network it falls over.

    H.

    Ill have a play with this tonight, (if the missus pemits it that is :)>>)
     
  9. hagesol

    Joined:
    Mar 26, 2007
    Messages:
    22
    Likes Received:
    0
    11:02:12.203 ClientRegs::eek:nSuccess Registration of sip:8539xxxx@sip.phonzo.com is successful

    11:02:12.000 ExtLine::Register_ Send registration for "105"<sip:8539xxxx@sip.phonzo.com>

    11:02:12.000 ExtLine::Register_ Use External IP for device line registration DN='10000' device='Phonzo'

    Like you see the VSP is register.... For me this is a big question,,, :)

    Lars
     
  10. Anonymous

    Anonymous Guest

    Damn, would be nice if it wasn't :). Ok back to the drawing board, perhaps it is one of the advanced settings. I have not had much time to play with that, so today might be the day :)
     
  11. hagesol

    Joined:
    Mar 26, 2007
    Messages:
    22
    Likes Received:
    0
    I have also a another sip-account I haven't used in 3CX, I registred the account in 3CX, but her I get this message:

    12:24:19.906 ExtLine::Register_ Send registration for "10001"<sip:xxxx@sip.vyke.com>

    12:24:19.906 ExtLine::Register_ Use External IP for device line registration DN='10001' device='vYKE'

    12:24:20.656 ClientRegs::eek:nFailure Registration of sip:xxxx@sip.vyke.com has failed; reason=Unauthorized

    Her is the settings for http://www.vyke.com:

    Settings for Vyke SIP:
    Display name: xxxxx
    Username: xxxxx
    Password: ***** (Same as your vyke password if confirmed)
    SIP server: sip.vyke.com
    IAX Server: iax.vyke.com
    Port: 5060
    RTP port: 5090
    DTMF: RFC 2833
    Proxy settings: Same as SIP server
    Register server: Leave blank
    Supported codecs: G.729, G.711 (ulaw), gsm

    I undestand I have written something wrong, but don't understand what,,,

    In the line status the light i red, and I understand why - the sip-account whan't registrer....

    Why i don't like this account is why this account don't have a phonenumber...

    lars
     
  12. hagesol

    Joined:
    Mar 26, 2007
    Messages:
    22
    Likes Received:
    0
    now I could call out with http://www.Vyke.com, it was wrong sip-adress,,,

    But don't with the http://www.phonzo.com who has phone number to recieve calls,,,

    This is soooooo sad for me,,,,

    Lars
     
Thread Status:
Not open for further replies.