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Flavio

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  1. Hello all. I don't know if it's the correct section to post, anyway: I'd like to know if there is the chance to get the standard/default PBX audio messages also spoken in other languages. Is there any location they could be downloaded from? Kind regards, F.
  2. Hello MR X! This is what I got in the "Call Log": 2011/04/14 11:29 My Secretary (42) Me Myself (41)(41) 00:27 0911234567 This is the corresponding "Logfile": [5] 2011/04/14 11:29:52: Identify trunk (line match) 1 [6] 2011/04/14 11:29:52: Sending RTP for qBdMwBMEmhK.qH4IvBtJPGsHjcIryJuq to 172.16.16.17:4000, codec not set yet [5] 2011/04/14 11:29:52: Global trunk 0911234567@pbx.piramide.local sends call to 72 in domain pbx.piramide.local [3] 2011/04/14 11:29:52: Hunt group 72 wants to add 4 members [6] 2011/04/14 11:29:52: Codec pcmu/8000 is chosen for call id qBdMwBMEmhK.qH4IvBtJPGsHjcIryJuq [6] 2011/04/14 11:29:59: Codec pcmu/8000 is chosen for call id fd630615@pbx [6] 2011/04/14 11:29:59: Sending RTP for fd630615@pbx to 172.16.16.231:52088, codec pcmu/8000 [5] 2011/04/14 11:29:59: INVITE Response 487 Request Terminated: Terminate 4ef1b0c7@pbx [5] 2011/04/14 11:29:59: INVITE Response 487 Request Terminated: Terminate 91b1e516@pbx [5] 2011/04/14 11:29:59: INVITE Response 487 Request Terminated: Terminate 5223328d@pbx [5] 2011/04/14 11:29:59: INVITE Response 487 Request Terminated: Terminate 773ca7ef@pbx [6] 2011/04/14 11:30:08: Codec pcmu/8000 is chosen for call id fd630615@pbx [6] 2011/04/14 11:30:09: Sending RTP for 3c6960323099-0iponi9rshnz to 172.16.16.231:64890, codec not set yet [6] 2011/04/14 11:30:09: Codec pcmu/8000 is chosen for call id 3c6960323099-0iponi9rshnz [6] 2011/04/14 11:30:13: Codec pcmu/8000 is chosen for call id 90b3ae3d@pbx [6] 2011/04/14 11:30:13: Sending RTP for 90b3ae3d@pbx to 172.16.16.230:61804, codec pcmu/8000 [5] 2011/04/14 11:30:19: Call fd630615@pbx: Last request not finished [6] 2011/04/14 11:30:19: Codec pcmu/8000 is chosen for call id 90b3ae3d@pbx [5] 2011/04/14 11:30:19: BYE Response: Terminate 3c6960323099-0iponi9rshnz [5] 2011/04/14 11:30:19: Tuning to new SSRC [5] 2011/04/14 11:30:19: BYE Response: Terminate fd630615@pbx [5] 2011/04/14 11:30:26: BYE Response: Terminate qBdMwBMEmhK.qH4IvBtJPGsHjcIryJuq Is it correct that the CallerID is missing for the incoming call? Thanks, F.
  3. Hello everybody. I noticed missing registrations in the "Call log". Following is the scenario: 1) external call coming in; 2) secretary (ext 42) picking up, talks to the remote part; 3) secretary transfers the call to my m9 by means of the BLF-keys (Function keys are configured with BLF for our extensions); 4) I answer the phone, talk to my secretary; 5) she hangs up; 6) call gets transferred; 7) I talk with the remote part; 8) call is terminated. If I then look at the "Call log", I just see the call between the two extensions, the external caller number does not appear! Is there any parameter I missed to set/configure? Thanks and kind regards, F.
  4. I've tried every single option but it didn't remove the IP address showing up. Again: may this depend on the VoIP provider's settings? Kind regards, F.
  5. So you are telling me that it should rather look like From: "Anonymous" <sip:USERNAME;user=phone>;tag=1204182133 If yes, is this depending on the VoIP provider or on my snomOne PBX? Thanks, F.
  6. Hello, back again. Here it goes: INVITE sip:46@192.168.1.220:1034;line=b0ac8vw4 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bK-a8348b8044f65b804662e8b425dafa9b;rport From: "Anonymous" <sip:192.192.192.192;user=phone>;tag=1204182133 To: <sip:41123456789@192.192.192.192;user=phone> Call-ID: fd24d171@pbx CSeq: 10787 INVITE Max-Forwards: 70 Contact: <sip:46@192.168.1.200:5060;transport=udp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: snom-PBX/4.2.0.3959 Alert-Info: <http://127.0.0.1/Bellcore-dr4> Content-Type: application/sdp Content-Length: 337 v=0 o=- 642798310 642798310 IN IP4 192.168.1.200 s=- c=IN IP4 192.168.1.200 t=0 0 m=audio 49768 RTP/AVP 8 0 9 2 3 101 a=rtpmap:8 pcma/8000 a=rtpmap:0 pcmu/8000 a=rtpmap:9 g722/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtcp-xr:rcvr-rtt=all voip-metrics a=sendrecv
  7. Hello everybody. When I'm out of office and nobody is there, I'd like to have my calls directly transferred to my cellphone. With traditional PSTN phones, I would do it with *210791234567, how may I achieve this same behaviour with snomOne PBX? Many thanks, F.
  8. Hello. I got this one now, from the Call log: [5] 2011/03/01 14:03:08: Identify trunk (line match) 1 [6] 2011/03/01 14:03:08: Sending RTP for bef1776-31dd02fe-73ed13a0-8112@62.65.128.50 to 195.190.166.249:54602, codec not set yet [5] 2011/03/01 14:03:08: Global trunk [12345678]@pbx.piramide.local sends call to 72 in domain pbx.piramide.local [3] 2011/03/01 14:03:08: Hunt group 72 wants to add 4 members [6] 2011/03/01 14:03:08: Codec pcmu/8000 is chosen for call id bef1776-31dd02fe-73ed13a0-8112@62.65.128.50 [5] 2011/03/01 14:03:22: INVITE Response 487 Request Terminated: Terminate 08914055@pbx [5] 2011/03/01 14:03:22: INVITE Response 487 Request Terminated: Terminate ee0bf877@pbx [5] 2011/03/01 14:03:22: INVITE Response 487 Request Terminated: Terminate e91673ea@pbx [5] 2011/03/01 14:03:22: INVITE Response 487 Request Terminated: Terminate bcc69cc1@pbx I didn't pick up the incoming call. F.
  9. Hello there! I recorder my IVR on my Auto Attendant (ext 70). I created ext 65 as a Service Flag account as follows: Mode: manual State: clean Confirmation tone: Service active/inactive Extensions that may change status: 41 Then I put 65 in the Auto Attendant IVR corresponding to my previously recorded message. If I put on my trunk the destination to ext 70 and I enable from ext 41 the Service Flag, I get to hear my message. What I now would like to do, is that first of all the phones extensions 40-43 should ring, then after say 20 seconds, the Service Flag should intervene. How to achieve this? Thanks, F.
  10. Where should I look for "from-header"? In the "Call Log" I see "From" = "Anonymous()" Kind regards, F.
  11. Hy there. I'd like to know how to set up some sort of "out-of-office" message, which I would like to be able to activate/deactivate manually. What's the best procedure to achieve this? I'm quite new to PBX in general, thus I'd need some help... Kind regards, F.
  12. Hello all. When an anonymous call (without Caller-ID) comes in, my snom 820 connected to snomOne shows the IP address of the sip provider. Is there any way to change this behaviour to have "anonymous" shown? Thanks, F.
  13. Hello all! I would like to know if there's a way to have a view of the call list as they come in (and go out) without having to log into the admin GUI of snomone. This would be of use for our users to be able to see what phone calls are coming in. Thanks for any suggestions. F.
  14. Hy there, is there any howto/guide to set up syslogd redirection? I'd like to configure it to log to localhost (OSX Snow Leopard Server) redirecting to a separate /var/log/snomOne.log file. Eventually, also intergrating some rotation would be nice. And, most important: which facility is used by snomOne to log? TIA.
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