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HedgeHog

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  1. OCS2007

    in Microsoft OCS

    Posted

    Maybe someone has a hint on a more OCS2007 specific problem.

     

    Now when we call a number in Communicator it rings two times before the real target-number phone rings. Then upon the first real ring at target phone communicator cancels with an error, that it does not received any audio from "number"....

     

    Looks like Communicator does not wait long enough for establishment of the real call?

     

    Foget this.... it was more a Firewallissue at a Branchoffice.

  2. OCS2007

    in Microsoft OCS

    Posted

    Hello again, pbxnsip does now everything as expected. Thx. And we will buy a license after evaluation is over.

     

    Maybe someone has a hint on a more OCS2007 specific problem.

     

    Now when we call a number in Communicator it rings two times before the real target-number phone rings. Then upon the first real ring at target phone communicator cancels with an error, that it does not received any audio from "number"....

     

    Looks like Communicator does not wait long enough for establishment of the real call?

  3. OCS2007

    in Microsoft OCS

    Posted

    Hehe. Then you need to set the password policy in admin mode in the settings to accept any password!

     

    A sorry.. got it.... just not seeing the forest between all those trees... Thank you!

  4. OCS2007

    in Microsoft OCS

    Posted

    You can clear the SIP password and put the IP address of the mediation server into the registration tab of the extension.

     

    I tried that, but pbxnsip config says "password is not secure enough" ?

  5. OCS2007

    in Microsoft OCS

    Posted

    Ok finaly I got it.

     

    Many thx again.

     

    Transfered pbxnsip to another machine, recreated config with localhost domain and at least it does what it should. :-)

     

    Many many thx.

     

    One last question. Is it somehow possible to create multiple SIP-Register-Accounts and use the tel:URI in OCS-Users to bring a communicator call out via pbxnsip on a specific account?

     

    Tried to create Accounts with the Primary name same as the TEL:URI. But I only get an access denied error. I think it is because SIP-From changes to internal Domain-Name when disabling "Assume that call comes from" to nothing.

     

    Log about this

     

    INVITE sip:01724025362@10.0.254.4;user=phone SIP/2.0

    FROM: <sip:+491805835684540@ops.internet.pc-soft.info;user=phone>;epid=D446D4C154;tag=3b5cd8562

    TO: <sip:01724025362@10.0.254.4;user=phone>

    CSEQ: 38 INVITE

    CALL-ID: 6b082cfa-bc4c-4d44-846a-749741eef94a

    MAX-FORWARDS: 70

    VIA: SIP/2.0/TCP 10.0.254.14:4859;branch=z9hG4bK46b5b52

    CONTACT: <sip:ops.internet.pc-soft.info:5060;transport=Tcp;maddr=10.0.254.14;ms-opaque=e6946a50e9b9afc2>

    CONTENT-LENGTH: 299

    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 10.0.254.14

    s=session

    c=IN IP4 10.0.254.14

    b=CT:1000

    t=0 0

    m=audio 63216 RTP/AVP 97 101 0 8

    c=IN IP4 10.0.254.14

    a=rtcp:63217

    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

     

    [7] 2007/11/05 15:53:05: UDP: Opening socket on port 64056

    [7] 2007/11/05 15:53:05: UDP: Opening socket on port 64057

    [5] 2007/11/05 15:53:05: Identify trunk (IP address and domain match) 6

    [9] 2007/11/05 15:53:05: Resolve destination 47: tcp 10.0.254.14 4859

    [7] 2007/11/05 15:53:05: SIP Tx tcp:10.0.254.14:4859:

    SIP/2.0 100 Trying

    Via: SIP/2.0/TCP 10.0.254.14:4859;branch=z9hG4bK46b5b52

    From: <sip:+491805835684540@ops.internet.pc-soft.info;user=phone>;epid=D446D4C154;tag=3b5cd8562

    To: <sip:01724025362@10.0.254.4;user=phone>;tag=add65be01d

    Call-ID: 6b082cfa-bc4c-4d44-846a-749741eef94a

    CSeq: 38 INVITE

    Content-Length: 0

     

     

    [9] 2007/11/05 15:53:05: Resolve destination 48: tcp 10.0.254.14 4859

    [7] 2007/11/05 15:53:05: SIP Tx tcp:10.0.254.14:4859:

    SIP/2.0 401 Authentication Required

    Via: SIP/2.0/TCP 10.0.254.14:4859;branch=z9hG4bK46b5b52

    From: <sip:+491805835684540@ops.internet.pc-soft.info;user=phone>;epid=D446D4C154;tag=3b5cd8562

    To: <sip:01724025362@10.0.254.4;user=phone>;tag=add65be01d

    Call-ID: 6b082cfa-bc4c-4d44-846a-749741eef94a

    CSeq: 38 INVITE

    User-Agent: pbxnsip-PBX/2.1.0.2115

    WWW-Authenticate: Digest realm="ops.internet.pc-soft.info",nonce="8000590fc939980dd38f090b01ca7883",domain="sip:01724025362@10.0.254.4;user=phone",algorithm=MD5

    Content-Length: 0

  6. OCS2007

    in Microsoft OCS

    Posted

    Hi,

     

    pbxsnip Version: 2.1.0.2115 (Win32)

     

    Assume that call comes from user= 491805835684540

     

    491805835684540 is Primary Name of an unused Account

     

    I also enhanced log to 0 and get this...

     

    SIP/2.0 100 Trying

    Via: SIP/2.0/TCP 10.0.254.15:3648;branch=z9hG4bKaecd8d52

    From: <sip:j.suenram@pc-soft.info>;epid=848AEC7FF1;tag=6228d44daf

    To: <sip:04445950215@10.0.254.15;user=phone>;tag=9f31276645

    Call-ID: 7d7a1e66-a33e-4650-9766-4e965f1a7c00

    CSeq: 24 INVITE

    Content-Length: 0

     

     

    [7] 2007/11/02 16:29:20: Set packet length to 20

    [6] 2007/11/02 16:29:20: Sending RTP for 7d7a1e66-a33e-4650-9766-4e965f1a7c00#9f31276645 to 10.0.254.15:61744

    [5] 2007/11/02 16:29:20: Received incoming call without trunk information and user has not been found

    [7] 2007/11/02 16:29:20: Set packet length to 20

    [9] 2007/11/02 16:29:20: Resolve destination 540: tcp 10.0.254.15 3648

    [7] 2007/11/02 16:29:20: SIP Tx tcp:10.0.254.15:3648:

    SIP/2.0 404 Not Found

    Via: SIP/2.0/TCP 10.0.254.15:3648;branch=z9hG4bKaecd8d52

    From: <sip:j.suenram@pc-soft.info>;epid=848AEC7FF1;tag=6228d44daf

    To: <sip:04445950215@10.0.254.15;user=phone>;tag=9f31276645

    Call-ID: 7d7a1e66-a33e-4650-9766-4e965f1a7c00

    CSeq: 24 INVITE

    Contact: <sip:04445950215@127.0.0.1:5065;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/2.1.0.2115

    Content-Length: 0

     

     

    [9] 2007/11/02 16:29:20: Resolve destination 541: tcp 10.0.254.15 3648

    [7] 2007/11/02 16:29:20: SIP Tx tcp:10.0.254.15:3648:

    SIP/2.0 404 Not Found

    Via: SIP/2.0/TCP 10.0.254.15:3648;branch=z9hG4bKaecd8d52

    From: <sip:j.suenram@pc-soft.info>;epid=848AEC7FF1;tag=6228d44daf

    To: <sip:04445950215@10.0.254.15;user=phone>;tag=9f31276645

    Call-ID: 7d7a1e66-a33e-4650-9766-4e965f1a7c00

    CSeq: 24 INVITE

    Contact: <sip:04445950215@127.0.0.1:5065;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/2.1.0.2115

    Content-Length: 0

     

     

    [7] 2007/11/02 16:29:20: SIP Rx tcp:10.0.254.15:3648:

    ACK sip:04445950215@10.0.254.15;user=phone SIP/2.0

    FROM: <sip:j.suenram@pc-soft.info>;tag=6228d44daf;epid=848AEC7FF1

    TO: <sip:04445950215@10.0.254.15;user=phone>;tag=9f31276645

    CSEQ: 24 ACK

    CALL-ID: 7d7a1e66-a33e-4650-9766-4e965f1a7c00

    MAX-FORWARDS: 70

    VIA: SIP/2.0/TCP 10.0.254.15:3648;branch=z9hG4bKaecd8d52

    CONTENT-LENGTH: 0

  7. OCS2007

    in Microsoft OCS

    Posted

    Hi again,

     

    I made everything as exact as you describe.

     

    And the call comes to the pbxsnip.

     

    But it logs.

    [5] 2007/11/02 11:25:42: SIP port accept from 10.0.254.15:2733

    [5] 2007/11/02 11:25:43: Received incoming call without trunk information and user has not been found

     

    Maybe you have another hint?

     

    Many thx!!!

  8. OCS2007

    in Microsoft OCS

    Posted

    HI!

     

    Yeah that looks very good and easy!

     

    Many many Thx!

     

    PS: Ich schulde dir was maaaan!

     

    Can you say if Voiping this way quality is OK? Does extra way for voip-data degrade quality?

  9. OCS2007

    in Microsoft OCS

    Posted

    Hi,

     

    maybe somone has already done this.

     

     

    OCS2007 <-> pbxnsip <-> SIP-Provider with simple Logon SIP-URI-Accounts.

     

    Can someone give hints or explain what exactly to configure in pbxsnip and in OCS2007, so that Communicator can make VoIP calls outbound?

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