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HedgeHog

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  1. Foget this.... it was more a Firewallissue at a Branchoffice.
  2. 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. A sorry.. got it.... just not seeing the forest between all those trees... Thank you!
  4. I tried that, but pbxnsip config says "password is not secure enough" ?
  5. 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. 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. 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. 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. HedgeHog

    OCS2007

    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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