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Andrep

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Posts posted by Andrep

  1. Disable PRACK

    in General Setup

    Posted

    Hi all,

     

    is it possible to disable the PRACK support?

     

    I have a gateway for OCS connectivity (OfficeMaster Gate) and i need to disable the PRACK support on the GW (their support asked me to do that for testing) because i have no audio between the GW and PBXNSIP with their new release.

    If I do so, the phone rings (a normal SIP-Phone connected to pbxnsip) only one time and then the call disappears.

     

    I just want to test what happens if i could disable the PRACK support on the pbxnsip...

     

    Thanks

  2. Dial out from OCS does not work

    in Microsoft OCS

    Posted

    Hi Jan,

     

    It works now. I had to replace the ip-address of the mediation server with the fqdn.

     

    Regarding the OfficeMaster Gateway I have still problems (no audio --> I guess codec mismatch). I'm in contact with their support and I will post the config as soon as the problems are solved.

     

    Hi Andrep,

     

    I assume that you have read my wiki page for Basic Setup for pbxnsip / Office Communications Server 2007 Interoperability OCS.

    If not, please read it very careful.

     

    I can see that the NOT FOUND is coming prompt from the pbxnsip. So I guess another trunk (outbound) is not involved, you need to check the trunk, configured to the OCS Meditiation Server again.

     

    Please compare your configuration with the one in the wiki. Like described, make sure that the Account under "Assume that calls come from user:" is existing. It must only exist, a registration is not necessary.

     

    btw.: I was trying to call your +41434435683 and it was succesfully ringing :D , but did no redirect to an Exchange 2007 Unified Messaging Server :D Simply Decline after some rings. Are you using the Ferrari Office Master now with pbxnsip for PSTN?

     

    You can check your call log, find my Berlin number and call me back if you like or simply look at my signature her ;)

     

    Best regards,

     

    Jan

  3. Dial out from OCS does not work

    in Microsoft OCS

    Posted

    Hi all,

     

    I configured PBXNSIP for OCS.

    I can dial in from a PSTN device to an extension and/or to the OCS, but i cannot dial out from OCS to a PSTN device (It works from an extension which is directly connected to PBXNSIP)

     

    I have two trunks configured (1 to OCS and 1 to the GW for PSTN connectivity)

     

    Here is the log (Communicator is +41434435683 and i want to dial +41763801234 which is a mobile phone):

     

    [5] 2008/06/03 15:23:13: SIP port accept from 172.25.20.66:4749

    [9] 2008/06/03 15:23:13: SIP Rx tcp:172.25.20.66:4749:

    INVITE sip:+41763801234@172.25.20.110;user=phone SIP/2.0

    FROM: <sip:+41434435683@mediation.collabcom.ch;user=phone>;epid=FF7797D3EA;tag=b42ca2ac31

    TO: <sip:+41763801234@172.25.20.110;user=phone>

    CSEQ: 13 INVITE

    CALL-ID: 5b978b49-1fdf-4745-94ba-ce4206d8d094

    MAX-FORWARDS: 70

    VIA: SIP/2.0/TCP 172.25.20.66:4749;branch=z9hG4bK7315eaa8

    CONTACT: <sip:MEDIATION.collabcom.ch:5060;transport=Tcp;maddr=172.25.20.66;ms-opaque=3d8ccc5b0ddbda89>

    CONTENT-LENGTH: 302

    SUPPORTED: 100rel

    USER-AGENT: RTCC/3.0.0.0 MediationServer

    CONTENT-TYPE: application/sdp; charset=utf-8

    ALLOW: UPDATE

    ALLOW: Ack, Cancel, Bye,Invite

     

    v=0

    o=- 0 0 IN IP4 172.25.20.66

    s=session

    c=IN IP4 172.25.20.66

    b=CT:1000

    t=0 0

    m=audio 55808 RTP/AVP 97 101 0 8

    c=IN IP4 172.25.20.66

    a=rtcp:55809

    a=label:Audio

    a=rtpmap:97 RED/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-16

    a=rtpmap:0 PCMU/8000

    a=rtpmap:8 PCMA/8000

    a=ptime:20

     

    [9] 2008/06/03 15:23:13: Resolve 5679: tcp 172.25.20.66 4749

    [9] 2008/06/03 15:23:13: SIP Tx tcp:172.25.20.66:4749:

    SIP/2.0 100 Trying

    Via: SIP/2.0/TCP 172.25.20.66:4749;branch=z9hG4bK7315eaa8

    From: <sip:+41434435683@mediation.collabcom.ch;user=phone>;epid=FF7797D3EA;tag=b42ca2ac31

    To: <sip:+41763801234@172.25.20.110;user=phone>;tag=3b5230389e

    Call-ID: 5b978b49-1fdf-4745-94ba-ce4206d8d094

    CSeq: 13 INVITE

    Content-Length: 0

     

     

    [9] 2008/06/03 15:23:13: Resolve 5680: tcp 172.25.20.66 4749

    [9] 2008/06/03 15:23:13: SIP Tx tcp:172.25.20.66:4749:

    SIP/2.0 404 Not Found

    Via: SIP/2.0/TCP 172.25.20.66:4749;branch=z9hG4bK7315eaa8

    From: <sip:+41434435683@mediation.collabcom.ch;user=phone>;epid=FF7797D3EA;tag=b42ca2ac31

    To: <sip:+41763801234@172.25.20.110;user=phone>;tag=3b5230389e

    Call-ID: 5b978b49-1fdf-4745-94ba-ce4206d8d094

    CSeq: 13 INVITE

    Contact: <sip:+41763801234@172.25.20.110:5060;transport=tcp>

    Supported: 100rel, replaces, norefersub

    Allow-Events: refer

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

    Accept: application/sdp

    User-Agent: pbxnsip-PBX/2.1.10.2474

    Content-Length: 0

     

     

    [9] 2008/06/03 15:23:13: Resolve 5681: tcp 172.25.20.66 4749

    [9] 2008/06/03 15:23:13: SIP Tx tcp:172.25.20.66:4749:

    SIP/2.0 404 Not Found

    Via: SIP/2.0/TCP 172.25.20.66:4749;branch=z9hG4bK7315eaa8

    From: <sip:+41434435683@mediation.collabcom.ch;user=phone>;epid=FF7797D3EA;tag=b42ca2ac31

    To: <sip:+41763801234@172.25.20.110;user=phone>;tag=3b5230389e

    Call-ID: 5b978b49-1fdf-4745-94ba-ce4206d8d094

    CSeq: 13 INVITE

    Contact: <sip:+41763801234@172.25.20.110:5060;transport=tcp>

    Supported: 100rel, replaces, norefersub

    Allow-Events: refer

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

    Accept: application/sdp

    User-Agent: pbxnsip-PBX/2.1.10.2474

    Content-Length: 0

     

     

    [9] 2008/06/03 15:23:13: SIP Rx tcp:172.25.20.66:4749:

    ACK sip:+41763801234@172.25.20.110;user=phone SIP/2.0

    FROM: <sip:+41434435683@mediation.collabcom.ch;user=phone>;tag=b42ca2ac31;epid=FF7797D3EA

    TO: <sip:+41763801234@172.25.20.110;user=phone>;tag=3b5230389e

    CSEQ: 13 ACK

    CALL-ID: 5b978b49-1fdf-4745-94ba-ce4206d8d094

    MAX-FORWARDS: 70

    VIA: SIP/2.0/TCP 172.25.20.66:4749;branch=z9hG4bK7315eaa8

    CONTENT-LENGTH: 0

     

     

    Thanks for your help. Could it be that my Dial Plan is not correct? I also need to replace the + to 00!

  4. OCS and Hunt Group

    in Microsoft OCS

    Posted

    Thanks a lot for the answers.

     

    Sounds great --> I will plan to deply a test installation of the pbxnsip.

     

    Just another question:

    Is it true that I will need to configure 2 trunks (1 to gateway and 1 to OCS)?

    In case I handle outgoing calls with OCS: do I need to create a trunk to the gateway for incoming calls?

  6. OCS and Hunt Group

    in Microsoft OCS

    Posted

    Hi all,

     

    We have an OCS and Exchange 2007 UC Setup. For PSTN Connectivity we use a Ferrari Electronic OfficeMaster Box.

    We are missing the hunt group feature on the Microsoft side and so we are in need of a PBX, which we not have at this moment...

     

    Does it work if i setup a hunt group in pbxnsip and add extensions of some OC-Clients?

    The goal is that our main phone number rings on multiple Office Communicator Clients

     

    Also i wanted to know if the ferrari officemaster gateway is supported with pbxnsip.

     

    If you are confident I will ask for an evaluation license and setup pbxnsip.

     

    Thanks a lot for your help

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