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Walter

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About Walter

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  1. Call drops after a minute

    Thanks I did send you a PM. I have only 1 Co Line because they have only 3 avail and 2 are in a different trunk to be able to send out different DID. Calls go out in 2 different Trunks for that reason. Incoming trunk will be never used probably.
  2. Call drops after a minute

    Hello I have a snomONE Plus with Broadvox as SIP trunk provider, and after 66 seconds the PBX asks Broadvox to reset and when broadvox does the PBX never replays back and we get a one sided audio only until the call drops all together. here is my Trunk Setup and a pcap from broadvox. Capture is too big to post here. I can send if you need it.
  3. All the sudden no calls can be made out. The sangoma error message is Yellow with this legend Failed to resolve hostname of "From" URI sip:200@pbx.xxxxx.com HELP
  4. Hot to divide 2 trunks from 2 Sangoma cards

    Everybody is part of the 2 companies, so I guess I'll need co lines, but I can't figure it out how the sangoma will identify this.
  5. Hello, I have a multi-company situation, using 2 sangoma cards right now, 4 Lines should go to 1 company and the other 4 to the other. Inbound is fine I set up the DID's on the Auto atendants,but Outband I can't find a way to make the calls going out wiht the right caller ID.
  6. DTMF problems & Sangoma Card

    Here is the second Auto Attendant. First one is basically pres 1 press 2 . It will go anywhere in the list but not if I dial an Extension while the recording is going. If you want send me a provate and i can give you IP and access if you want to take a look at the PBX
  7. DTMF problems & Sangoma Card

    No it doesn't that is the problem, it seams to get the dtmf but it doesn't go any where. I added the trunk to my prior post probably after you replied to it
  8. DTMF problems & Sangoma Card

    OK I'm remote so I can't get to the Office phones right now. But here is the inbound call log. I'm also attaching the Trunk setup. Note that I already tried with the Out of band DTMF [8] 2012/01/28 08:55:49: Last message repeated 34 times [9] 2012/01/28 08:55:49: UDP: Opening socket on 0.0.0.0:64064 [9] 2012/01/28 08:55:49: UDP: Opening socket on 0.0.0.0:64065 [9] 2012/01/28 08:55:49: UDP: Opening socket on [::]:64064 [9] 2012/01/28 08:55:49: UDP: Opening socket on [::]:64065 [9] 2012/01/28 08:55:49: Resolve 19925: aaaa udp 127.0.0.1 5066 [9] 2012/01/28 08:55:49: Resolve 19925: a udp 127.0.0.1 5066 [9] 2012/01/28 08:55:49: Resolve 19925: udp 127.0.0.1 5066 [7] 2012/01/28 08:55:49: Set packet length to 20 [6] 2012/01/28 08:55:49: Sending RTP for 2a51371a-1dd2-11b2-8cc6-fbcd0b20cc98@S1-MYPBX to 127.0.0.1:14006, codec not set yet [8] 2012/01/28 08:55:49: Incoming call: Request URI sip:2148281088@localhost:5060;transport=udp;user=phone, To is "2148281088" <sip:2148281088@localhost:5060> [8] 2012/01/28 08:55:49: Trunk Netborder Express@pbx.MYPBX.com has country code not set, area code not set [9] 2012/01/28 08:55:49: Incoming: formatted From is = "GRANDPRARI TX" <sip:12144486474@127.0.0.1:5066;user=phone> [9] 2012/01/28 08:55:49: Incoming: formatted To is = "2148281088" <sip:2148281088@localhost:5060;user=phone> [9] 2012/01/28 08:55:49: Incoming: formatted URI is = sip:2148281088@pbx.MYPBX.com:5060;transport=udp;user=phone [8] 2012/01/28 08:55:49: Set the To domain based on To user 401@pbx.MYPBX.com [7] 2012/01/28 08:55:49: set_codecs: for 2a51371a-1dd2-11b2-8cc6-fbcd0b20cc98@S1-MYPBX codecs "0", codec_preference count 2 [8] 2012/01/28 08:55:49: Play audio_moh/noise.wav [8] 2012/01/28 08:55:49: call port 30: state code from 0 to 183 [7] 2012/01/28 08:55:49: Set packet length to 20 [9] 2012/01/28 08:55:49: update_codecs for 2a51371a-1dd2-11b2-8cc6-fbcd0b20cc98@S1-MYPBX: adding codec pcmu/8000 to available list [9] 2012/01/28 08:55:49: update_codecs for 2a51371a-1dd2-11b2-8cc6-fbcd0b20cc98@S1-MYPBX: codec_preference size 2, available codecs size 2 [6] 2012/01/28 08:55:49: Codec pcmu/8000 is chosen for call id 2a51371a-1dd2-11b2-8cc6-fbcd0b20cc98@S1-MYPBX [9] 2012/01/28 08:55:49: Resolve 19926: aaaa udp 127.0.0.1 5066 [9] 2012/01/28 08:55:49: Resolve 19926: a udp 127.0.0.1 5066 [9] 2012/01/28 08:55:49: Resolve 19926: udp 127.0.0.1 5066 [9] 2012/01/28 08:55:49: Resolve 19927: aaaa udp 127.0.0.1 5066 [9] 2012/01/28 08:55:49: Resolve 19927: a udp 127.0.0.1 5066 [9] 2012/01/28 08:55:49: Resolve 19927: udp 127.0.0.1 5066 [8] 2012/01/28 08:55:49: Packet authenticated by transport layer [8] 2012/01/28 08:55:50: Play recordings/att110.wav space20 [8] 2012/01/28 08:55:50: call port 30: state code from 183 to 200 [9] 2012/01/28 08:55:50: Resolve 19929: aaaa udp 127.0.0.1 5066 [9] 2012/01/28 08:55:50: Resolve 19929: a udp 127.0.0.1 5066 [9] 2012/01/28 08:55:50: Resolve 19929: udp 127.0.0.1 5066 [8] 2012/01/28 08:55:55: Packet authenticated by transport layer [8] 2012/01/28 08:56:00: Last message repeated 4 times [7] 2012/01/28 08:56:00: Received RFC4733 DTMF on codec 101 [6] 2012/01/28 08:56:00: Received DTMF 2 [8] 2012/01/28 08:56:00: call port 30: state code from 200 to 200 [8] 2012/01/28 08:56:00: Play audio_moh/noise.wav [6] 2012/01/28 08:56:01: Received DTMF 1 [8] 2012/01/28 08:56:01: Packet authenticated by transport layer [6] 2012/01/28 08:56:01: Received DTMF 2 [8] 2012/01/28 08:56:03: Packet authenticated by transport layer
  9. Hello, recently installed a Snom one plus with 2 sangoma cards on it. I can receive and make calls fine. Some of the DTMF work, like when pressing 1 2 and so, (just one digit. but if they dial an extension it wont go anywhere. Here is the Log file of 2 different calls. Log 1 INFO: - Called into Attendant, and dialed extension 212. Nothing happened, I hanged up. - 401 is main Attendant - 212 is an extension. -Attached image is the settings on 401 Attendant -------------------------------- LOG 1 ------------------------------------------ [5] 2012/01/27 16:12:43: Identify trunk (IP address and DID match) 1 [7] 2012/01/27 16:12:43: Set packet length to 20 [6] 2012/01/27 16:12:43: Sending RTP for 08cd5ede-1dd2-11b2-b7ac-97770de55725@S1-XXXXX to 127.0.0.1:14182, codec not set yet [5] 2012/01/27 16:12:43: Domain trunk Netborder Express@pbx.XXXXX.com sends call to 401 in domain pbx.XXXXX.com [7] 2012/01/27 16:12:43: set_codecs: for 08cd5ede-1dd2-11b2-b7ac-97770de55725@S1-Sliquid codecs "0", codec_preference count 2 [7] 2012/01/27 16:12:43: Set packet length to 20 [6] 2012/01/27 16:12:43: Codec pcmu/8000 is chosen for call id 08cd5ede-1dd2-11b2-b7ac-97770de55725@S1-XXXXX [7] 2012/01/27 16:12:54: Received RFC4733 DTMF on codec 101 [6] 2012/01/27 16:12:54: Received DTMF 2 [6] 2012/01/27 16:12:55: Received DTMF 1 [6] 2012/01/27 16:12:55: Received DTMF 2 --------------------------------END Log 1 --------------------------------------- Log 2 INFO: (Just 1 digit option works fine) - Called into attendant, selected Option 2 (secondary Attendant), then SELECTED option 3 (Agent Group) and it took me to a voicemail since it is the normal when no agents are logged on. ------------------------------ LOG 2 ------------------------------------------- [5] 2012/01/27 16:30:32: Identify trunk (IP address and DID match) 1 [7] 2012/01/27 16:30:32: Set packet length to 20 [6] 2012/01/27 16:30:32: Sending RTP for 862e1dd0-1dd2-11b2-af4c-fe9c9b9c9ee8@S1-Sliquid to 127.0.0.1:14188, codec not set yet [5] 2012/01/27 16:30:32: Domain trunk Netborder Express@pbx.xxxxx.com sends call to 401 in domain pbx.xxxxx.com [7] 2012/01/27 16:30:32: set_codecs: for 862e1dd0-1dd2-11b2-af4c-fe9c9b9c9ee8@S1-Sliquid codecs "0", codec_preference count 2 [7] 2012/01/27 16:30:32: Set packet length to 20 [6] 2012/01/27 16:30:32: Codec pcmu/8000 is chosen for call id 862e1dd0-1dd2-11b2-af4c-fe9c9b9c9ee8@S1-xxxxx [7] 2012/01/27 16:30:44: Received RFC4733 DTMF on codec 101 [6] 2012/01/27 16:30:44: Received DTMF 2 [6] 2012/01/27 16:31:07: Received DTMF 3 [6] 2012/01/27 16:31:07: ACD 601: Set priority to 1 [7] 2012/01/27 16:31:07: set_codecs: for 2c8d6786@pbx codecs "", codec_preference count 7 [7] 2012/01/27 16:31:08: Call 2c8d6786@pbx: Clear last request [7] 2012/01/27 16:31:27: Last message repeated 2 times [7] 2012/01/27 16:31:27: Call 2c8d6786@pbx: Clear last INVITE [5] 2012/01/27 16:31:27: INVITE Response 487 Request Terminated: Terminate 2c8d6786@pbx [6] 2012/01/27 16:31:35: Reg 156, Sent MWI notification to user 200@pbx.xxxxx.com -----------------------------END Log 2 ------------------------------------------- Please HELP, thank you.
  10. I can't find the way to setup a Static IP on the snom one plus. Also can't decide if is better to use the Lan port or Wan port. I will have phones outside the office, just 2 but I guess WAN will be correct unless I stablish a VPN and use LAN. correct? still can't setup a static ip on ether of them. HELP
  11. Licensing

    Hello any body out there?
  12. Licensing

    The member snom ONE cannot receive any new messages Can you send me a PM with your User? same thing happens with pbx_support
  13. Licensing

    Can't it says snom one is a groupor something like that. pbx_support? in here or @snom.com?
  14. Licensing

    Still not working where should I send the key? Won't let me send to snom one Hello, thank you for your answer. I have a cable on the wan plier to my router (privaye IP) and another for the internal LAN and is still not activating.
  15. Licensing

    I just started a new snom ONE, I got the License on the back of the book that is 12 characters long xxx-xxx-xxx-xxx (groups of 3) the Help shows groups of 4, it Reads Activation Key, but every time I try to activate it wont do it. Does it need to be connected to the internet to activate? It doesn't say that any where, I'm pre-configuring it, so didn't think it will be necessary. I'm trying to get it ready for monday Ideas?
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